Posts from ‘Small Medium Business in Canada’
Asterisk + Skype + SMB = Freetalk Connect
Jazinga and Freetalk have combined efforts and the result is a Skype enabled SMB phone system called Freetalk Connect.
The press release:
FREETALK Partners With Jazinga To Create FREETALK® Connect
Companies Collaborate On Skype-enabled Small Business Communication System
Featuring Set Up In Less Than 15 MinutesMIAMI, January 20, 2010 — As the result of a new partnership announced today at ITEXPO East 2010, FREETALK and Jazinga have created the FREETALK® Connect, a full-featured unified communications system that is the first to feature Skype for SIP and Skype for Asterisk functionality.
FREETALK and Jazinga collaborated in designing the FREETALK Connect, featuring a do-it-yourself (DIY) technology approach that can be configured in less than 15 minutes, enabling users who are not tech savvy to use it without formal training. This new class of DIY communications system allows anyone with basic knowledge of computers to install and maintain the office phone system. SIP, Skype and traditional PSTN phones can be plugged into the network, and the FREETALK Connect auto-detects and configures them. An onscreen wizard guides the user through setup. Adding users and administering the system after install is equally simple.
Further distinguishing the FREETALK Connect is its intelligent routing capabilities. Incoming Skype calls, as well as SIP, PSTN and IAX2 calls, can be routed to any local or remote Skype user, SIP, analog or mobile phone. Additionally, the FREETALK Connect enables users to set up “Find Me, Follow Me” features, and provides a unified mail box that consolidates messages from voice mail and email into one mailbox.
Some of the key features from the Jazinga platform found in the FREETALK Connect include:
Callback / Dial-around
Access to Skype Buddy lists
Auto Attendant / IVR
Paging
Call Parking
Remote Extensions
Music on Hold
ConferencingThe FREETALK Connect also has an easily configured and updated:
Managing routes to users, telephone services, and applications
Providing SIP/Skype telephone service management
Router management (networking, port forwarding, DNS, DHCP)“Jazinga’s products consistently ensure call integrity by integrating quality of service and prioritizing voice traffic on the network into an affordable, simple product,” said In Store Solutions COO Craig Smith. “There was no question that FREETALK wanted to partner with Jazinga to develop the FREETALK Connect, because it continues our goal of working with the best providers to distribute outstanding products around the world.”
“FREETALK Connect is designed for small businesses with between 2 and 49 users, an undersold market that desperately needs UC functionality,” said Randy Busch, CEO of Jazinga Inc. “As a result of our partnership with In Store Solutions, the telecom technology playing field is much more level between larger enterprises and their smaller competitors.”The the FREETALK Connect is marketed through Skype Shop, which is operated by In Store Solutions. The unit initially will be available to registered U.S. Skype users beginning in March.
For more information about FREETALK Connect PBX or to order a unit, visit
http://freetalkconnect.com.
About FREETALK
FREETALK is a product innovation catalyst – identifying market gaps and working with its global partners to design, manufacture and quickly bring to market products that disrupt traditional categories. Leveraging untapped market opportunities, FREETALK products are designed to be environmentally friendly, sold online and delivered globally at aggressive price-points. Always at the forefront of innovation, FREETALK is known for creating synergistic products that add unique value to its partners’ branded points-of-sale.
About Jazinga
Jazinga Inc. develops communications products for small businesses and homes. The Jazinga system provides enterprise telephony and data functionality for this market, but at a fraction of the cost and without the setup complexity of an enterprise-class IP PBX. Jazinga Inc. is privately held and headquartered in Toronto, Canada. Additional information is available at www.jazinga.com.
Contact:
Sue Huss, for In Store Solutions
sue.huss@comunicano.com
+1 619-379-4396
Jazinga came to market a while back with a Asterisk appliance that is not much different than other you would find in the Asterisk market today. Skype recently announced their Skype SIP Trunking capability which is helping Skype become more open standards compliant, paving the way for deals like this one.
Since I have not tested the system myself I can only speculate that it is not huge departure from other Asterisk systems, which are not trivial to set up. Let’s hope they did their homework and come to market (March) with something that is much less technical and more end-user friendly, like Response Point.. was.
One thing that I find interesting is that it will be sold via the Skype store to US registered Skype users. If you were wondering what the connection is between Freetalk and Skype; the creators of Freetalk are also the curators of the Skype store. Ya, you heard me right. The company that created Freetalk (In Store Solutions) operates the Skype store. Which makes one wonder if there is overlapping ownership between Skype and In Store Solutions.
Something else that I find interesting, and not just because I am one of the founders of Xten/Counterpath, is how this announcement relates the recent announcement of the Asterisk/Digium softphone from Counterpath. Which may be why In Store Solutions decided not to leverage the Digium or Asterisk brand in this release, maybe they see the new Asterisk Bria softphone as a competitor in this instance?
I expect this will not be the last Asterisk-based phone system to incorporate Skype functionality this year, but it would seem as though they are the first, congrats to fellow Canadians at Jazinga.
Written by Erik Lagerway - Visit Website
The days are numbered for all Free Conference Call services, it’s simply a matter of time. The big telcos have been a bit pissy for having to aid their competitors indirectly via the USF. The emotion over this has been coming to boil for years now and recently Free Conference Call provider Foonz fell, just a few days ago.
I am sure glad we decided to pull out of that Free Conference Call game long ago. Our conference call service “Lypp” (formerly Gaboogie) started by offering free conferencing but quickly decide that was a bad idea (duh!). Lypp is now cash flow positive, growing like crazy and not showing any signs of slowing down.
Written by Erik Lagerway - Visit Website
SIP trunks are simply another way of saying VoIP Provider for your phone system. A SIP trunk is a connection from a PBX (phone system) using SIP (Session Initiation Protocol) to an ITSP (Internet Telephony Service Provider).
It might sound complicated but it’s really quite simple, SIP trunks take the place of your legacy telephone company. Most phone systems out there today are more than a couple of years old and are likely based on circuit switched technology. Newer IP-PBXs use packet switching technology, which means they leverage the Internet to deliver the same features you have now, and then some. The difference could be minor or major depending on what your PBX is capable of and what your ITSP can deliver in terms of features and functionality.
Since the PSTN (public switch telephone network) is tied to aging circuit switched technology it has limitations in terms of what media it can support. Essentially, it can deliver low quality voice, that’s it.
SIP Trunks replace older PRI and POTS interfaces that we are used to and bring to the table a wide variety of communications options. Depending on your IP-PBX and your ITSP you could potentially look forward to HD (High Defenition) Voice and potentially HD Video.
HD voice (and video) for small business in Canada will happen, it’s only a matter of time. As broadband providers increase upstream bandwidth and dual WAN link-failover devices become common place, SIP trunking will accellerate in growth and on-net (calls made on the ITSP network) HD Voice will become common place.
Unfortunately, HD communication off-net (eg. PSTN) is not going anywhere at any great speed. Jeff Pulver is back as he reboots the communications industry with his new HD Communication Summit. I welcome Jeff back with open arms, if anyone can convince operators to increase speed towards wide-band/HD adoption it would Jeff Pulver.
Today we can see SIP trunking providers and hosted pbx providers supporting wideband codecs and devices on their networks. This will allow user to communicate in high definition with other users that have devices that support it, in brief you could have better calls between you and your colleagues in the office and remote office workers connected to the same PBX, and that is a step in the right direction.
Written by Erik Lagerway - Visit Website
G.711 is the default audio CODEC for most Response Point phones and requires approximately 90Kbps bandwidth upstream (your voice going out) and 90Kbps bandwidth downstream (your caller’s voice coming in).
To calculate peak usage take the peak concurrent callers x 90Kbps. For example: 5 concurrent calls x 90Kbps = 450Kbps is the required bandwidth for each direction. Keep in mind, this does not account for VPN usage for remote users or voice mail to email etc.
As an example, if you have a 1Mbps ADSL connection from your service provider, you might have an upstream bandwidth of approximately 700 Kbps. A conservative approach is to estimate just over half of the upstream bandwidth is available, ISPs generally over-sell their bandwidth. In this case, you could safely support 4 simultaneous G.711 calls if you were not doing anything else (e.g. downloading email, listening to online radio, downloading large files, etc.) on that connection.
The SMB Digital Voice network also supports G.729, which uses approximately 20Kbps bandwidth upstream (your voice going out) and 20Kbps bandwidth downstream (your caller’s voice coming in) for each call. G.729 provides very good call quality while minimizing bandwidth usage. The only noticeable difference would likely arise during on-net calls (calling other users on the SMB Phone network). G.711 offers a higher quality on-net call because G.711 does not compress audio, but as soon as the the call is handed off to the PSTN the call quality between G.711 and G.729 is hardly noticeable.
G.729 offers some real benefits, the most obvious is the 400% decrease in bandwidth capacity requirements. G.729 also handles Jitter more efficiency during times where low bandwidth / high congestion would likely render a similar call using G.711 unintelligible.
You can force your phone to use G.729 on Response Point handsets but some are harder to configure than others. For example, on Aastra 675x phones the global SIP settings are grayed out out via Javascript on page load making it tough to set the codec.
As a general rule of thumb, we like to recommend an independent broadband connection that you can use for Response Point. You may want to acquire a router that has dual WAN link failover, VPN Server (for remote sites) and some QOS traffic shaping functionality.
Written by Erik Lagerway - Visit WebsiteI wrote an article over at the SMB Phone blog on Response Point VPNs and remote workers. If you are having some issues with VPNs and Response Point this might help.
Written by Erik Lagerway - Visit Website
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Back in 2004 I wrote a post relating to the VON Canada Panel I sat on with Niklas Zennstrom. It was an interesting debate on open standards (SIP in this case) and closed networks, specifically Skype. I was quite vocal about how silly I thought Skype was not to include SIP, a few of you picked up on that 
It looks like something good came of the eBay purchase as we now see a Skype pushing towards open standards, good stuff!
On a similar note, I heard a rumour that it’s likely Jason Fischl the current CTO at Counterpath (Xten) will be going over to work with Jonathan Christensen (General Manager – Media Platform) at Skype. Jason was an early advocate of SIP in the IETF and works with some of the best minds on the subject: Cullen Jennings, Robert Sparks, Alan Duric come to mind.
This could get interesting.
I will do some testing with SkypeforSIP & Response Point and post the results along with my comments on what this new offer from Skype might mean for Response Point.
Written by Erik Lagerway - Visit WebsiteSubscribe
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