ITEXPO, Booth 901 (Miami)-February 1, 2012-Digium®, Inc., the Asterisk® Company, today introduced a new family of high-definition Digium IP phones. They are the first that are engineered to fully leverage the power of Asterisk, the world’s most widely adopted open source communications software, and Switchvox. With Digium technology on both the server and the phone, users will benefit from the best possible performance, unprecedented integration and a uniquely customizable phone system.

“Digium’s new phones mark the launch of the next chapter in our history of innovation,” said Danny Windham, president and CEO of Digium. “These are the first phones designed specifically for Asterisk-with the tightest integration possible between the phones and Asterisk. The success of Asterisk began with the transfer of power from the hands of the proprietary phone vendors to the hands of end users and administrators of phone systems. And now we’ve done it again by bringing control to the desk phone. These phones are absolutely the easiest to install, integrate, provision and use with both Asterisk and Switchvox. And best of all, we’ve done all of this at a very competitive price point, providing our customers with the best value in business phone systems.”

Asterisk has always been about flexibility, allowing integrators and developers to create highly customized solutions. Likewise, Digium phones include an app engine with a simple yet powerful JavaScript API that lets programmers create custom apps that run on the phones. They aren’t simply XML pages; Digium phone apps can interface directly with core phone features.

“The app engine is a game-changing feature that will allow developers to write their own apps that run on the phones,” explained Mark Spencer, founder and chief technology officer for Digium. “We have a community of more than 80,000 users and developers who create amazing things with Asterisk. I look forward to seeing the cool apps they will create with these innovative phones. As usual, we’re enabling developers to create solutions limited only by their imaginations.”

Digium has leveraged this unique programming interface of the phones to create a suite of productivity applications that work with both Asterisk and Switchvox. Switchvox includes a unique web interface called Switchboard that gives each system user control of their personal communications environment. Digium has extended the capabilities of the Switchboard to the phone, putting advanced features like presence management, searchable contact directory, queue monitoring, recording and voicemail control, all at the user’s fingertips.

“We’ve had the opportunity to work with the phones early on, and for the first time, we’re using phones truly designed for Asterisk,” said Chris Green of Bema Information Technology. “Digium has delivered phones that simplify the most time-consuming portion of a system deployment-installation and provisioning. The phones automatically locate the server and assign the correct users, making our deployments faster than ever before. Additionally, the tight integration with Switchvox offers our customers an awesome user experience that’s just not possible with other phones.”

The Digium phones include the following models:

• The Digium D40 IP Phone -An entry-level HD IP phone with 2-line keys. This is Digium’s best value phone, designed for any employee in the company.
• The Digium D50 IP Phone -A mid-level HD IP phone with 4-line keys and 10 rapid dial/busy lamp field (BLF) keys with an easy to print paper label strip for the user’s most important contacts. This model is perfect for users who spend a lot of time on the phone.
• The Digium D70 IP Phone -An executive-level HD IP phone with 6-line keys and 10 rapid dial/busy lamp field (BLF) keys and real-time status information displayed on an additional LCD screen, allowing users to quickly navigate through up to 100 of their most important contacts. Designed for administrators or executives, the D70 offers top-of-the-line features.

The VXi UC ProSet is a step up in style and comfort, making it perfect for use in contact centers, at-home offices and other business settings. Designed to be lightweight and deliver crisp, natural audio, the UC ProSet also blocks 87 percent of all background noise.

said President and CEO Michael Ferguson of VXi Corporation.

VXi also state

Here are some of the main features that are available in the VXi UC ProSet headsets.

Features

• Wideband audio for clearer conversations and greater productivity.
• Superior noise canceling eliminates background noise and improves communication.
• UC-ready to fit right into your Unified Communications strategy.
• Lightweight, attractive design is easy and comfortable to wear, all day long.
• High-quality durability stands up to all-day, every-day use.
• Interchangeability with other headset brands lets you maximize your existing equipment investment.
• Two models to choose from, binaural (two speakers) and monaural (one speaker).

The VXi UC ProSet delivers a host of real-world business benefits.

• More productive conversations. Wideband audio and advanced noise canceling provide consistently clear, natural sound.
• More focused users. Smart, ergonomic design is comfortable and stays put, so users spend time working, not constantly adjusting their headsets.
• More efficient communications. Leading-edge technology maximizes Unified Communications potential.
• Better all-around value. High quality materials and construction means it’s built to last and last, even when used all day, every day.
• Best possible return on your investment. Compatibility with VXi, Plantronics and GN Netcom®/Jabra equipment means you can keep on using your existing amplifiers and direct connect cords.

Your Digium Switchvox SMB customers will be able to access their voicemail, check their call history, change out-of-office greetings and make calls from Switchvox through Switchvox Mobile. It makes it easy to place calls from their Switchvox extension and then transfer the call or record the call through the InCall Menu. Contact Us to find out more about Switchvox Mobile and how to extend the power of your Switchvox solution on Android mobile devices.

	Make Calls Call your clients or your coworkers from Switchvox Mobile, and the call looks like it’s coming from your desk!
	Check Voicemail Listen to your voicemail and return calls right from the Voicemail screen.
	Check Calls Call Logs show your incoming, outgoing, and missed calls. You can easily return calls right from the Call Log.
	Manage Greetings You can choose which of your voicemail greetings your callers hear.
	Company Directory The Directory lets you quickly find your coworkers’ extensions and email addresses, then call or send an email. iPhone Activate Call Rules
	Activate Call Rules Activate your call rules right from your smartphone. If you’re in an offsite meeting and can’t be disturbed, it’s easy to send your calls to voicemail.

As a provider of telecoms services, you have to supply your customers with great products as well as great service. This means finding terminals that impress and delight your customers, whilst still being reliable and easy to configure and support.

At Panasonic, a great deal of thought has gone into ensuring that both end users and telecoms service providers have a lot to like in the KX-UT series of SIP terminals.

Your customers will love the best in class sound quality, ease of use and low power consumption; while you will be impressed with the quality and versatility of the available configuration options – from local configuration via the built-in web server, through to completely automated bulk setup using a Panasonic redirection server delivering fully automatic ‘hands free’ configuration. Your customers just need to “plug in and turn on”.

Provisioning

Supplying telecoms services is a global business, and Telephony Service Providers (TSPs) are expected to support equipment wherever it is deployed. With the KX-UT series of SIP terminals, Panasonic has gone to great lengths to make deployment and support as easy as possible, whatever your customers location.

The Panasonic KX-UT series enables no fewer than three types of provisioning, with various degrees of automation, giving you all the options you need to provide premium service levels to your customers. Panasonic’s remote administration and automatic configuration technologies are designed to minimise project roll-out time, reduce TSP costs and improve the user experience.

As a service provider you can choose the configuration system that best meets your needs:

• Redirection Server – All the terminals ship with the address of the redirection server built-in. At power up, the phones contact the redirection server to identify their provisioning server. In turn, the provisioning server supplies complete configuration data to the phone. The customer simply – plugs in, turns on – and receives service once configuration completes.
• Provisioning Server with preconfigured URL – For high volume TSPs with their own provisioning server, Panasonic can pre-configure the phones with a provisioning server address.
• DHCP option 66 (RFC2132) – For larger, local installations, DHCP option 66 can be used to provide the address of a TFTP server which provides configuration files to the phones.

In order to ensure data security, HTTPS server certificates are supported, to allow secure transfer and connection to the provisioning server.

Whichever provisioning system you choose, your customers’ systems can be up and running securely, accurately and quickly.

Panasonic KX-UT series SIP terminals – Make yourself heard.

For more information on the Panasonic KX-UT SIP Phone Terminal range, please visit the respective models’ product page:

• Panasonic KX-UT113 SIP Phone Terminal
• Panasonic KX-UT123 SIP Phone Terminal
• Panasonic KX-UT133 SIP Phone Terminal
• Panasonic KX-UT136 SIP Phone Terminal
• Panasonic KX-UT248 SIP Phone Terminal
• Panasonic KX-UT670 SIP Phone Terminal

Panasonic KX-UT SIP Phones Comparison Table

	KX-UT113	KX-UT123	KX-UT133	KX-UT136	KX-UT248	KX-UT670*
Display
LCD Display	Monochrome Graphical				Greyscale Graphical	Colour 256k
LCD Size	242 x 55 pixels – 3 lines			242 x 109 pixels – 6 lines	4.4 inch	7 inch touch
LCD Contrast	6 levels					No
LCD Backlight	None	On/Off/Auto				15/Auto
HD Video	No					720p
Install Options
Desk Mount Tilt	No		Yes – 2 positions
Wall Mount	KX-A432 (optional)		KX-A433 (optional)			KX-A434 (optional)
Power Adaptor	KX-A239 (optional)				KX-A422 (optional)
Networking
Ethernet Ports	1 – 10/100	2 – 10/100			2 – 10/100/1000
Power Over Ethernet	Yes
Bluetooth	No				Yes	Optional
Audio Features
Handset, Speaker, Headset Voume	8 levels (includes echo cancellation and distortion prevention)					15 levels
Ringtones	32
Ringer Volume	6 levels + Off					7 levels + Off
Headset Port	2.5mm
Electronic Hook Switch Control Port	None		3.5mm Plantronics Compatible
Audio Codec	G.711, G.722, G.726, G.729a					G.711, G.722, G.729a
Keys
Soft Keys	Yes – 4					Yes – Screen
Programmable Keys	0		24		24 ( 3 pages of 8 )	24 ( 4 pages of 6 )
0-9, *, # Keys	Yes					Yes – Screen Keys
Nav & Cancel Keys	Yes					On Screen + Menu Keys
Software Features
Phone Book (Entries)	100 – 5 numbers each	500 – 5 numbers each				Depends on Memory**
Call Log Entries	None	30 incoming + 30 outgoing				100 in + 100 out
Conferencing	3 parties with terminal – multi party dependent on server
XML	XML Application Support from Q1 2012
Music On Hold	Supplied by host service (PBX/SIP Server)
IP Features
SIP Accounts	2		4		6
SIP Compatibility	RFC 3261 Standard SIP Server, Asterisk, Broadsoft, Panasonic SIP PBX
IP Version	IPv4
DHCP Client	Yes
DNS	Yes
HTTP	Yes
HTTPS	Yes
SNTP Client	Yes
VLAN (802.1q)	Yes
QoS (DiffServ)	Yes
802.1x	No				Yes
Configuration
Plug & Play Config.	Server based configuration, TR-069, Panasonic Redirect Server
Manual Config.	Internal with Configurator, Local (LCD based) network configuration

* – Preliminary Specifications

** – Up to 500 connected to NS1000

The first 5-Megapixel camera in the GXV product family, the GXV3651_FHD adds another layer to Grandstream’s solutions options by delivering the utmost resolution with exceptional low light sensitivity and functionality at a staggeringly attractive price. It’s now easier to capture hard-to-see image details with extreme high quality under various light conditions such as employees counting money or players’ activities in casinos, face recognition in high traffic areas such as airports or train stations, money transactions in various bank teller windows and car plate recognition at busy toll booths or in a large parking lot, for example.

Features

• Cutting edge H.264 real-time video compression (D1 resolution) with excellent image clarity at low-to-modest bandwidth
• SIP/VoIP for full duplex 2-way audio and video streaming to mobile phones and video phones
• Integrated PoE, embedded analytics, large pre-/post-event recording buffer
• High quality 5 Megapixels CMOS sensor and lens to ensure razor sharp picture quality
• Advanced multi-streaming-rate real-time H.264, Motion JPEG at 1080p resolution
• Alarm input & output, audio input & output, RS485
• 24MB pre-/post-event recording buffer
• USB 2.0 port and SD slot for storage/wireless network card
• Integrated Power-over-Ethernet (802.3af), optional antenna for wireless network
• Embedded video analytics, SIP/VoIP support for full duplex two-way audio as well as video streaming to mobile phones and video phones
• Built-in high performance streaming server to allow up to 16 simultaneous D1 viewers or 25 CIF viewers
• Combine with GSurf software to support remote viewing, controlling and recording of up to 36 cameras from anywhere on the Internet or corporate network.

Technical Specification

• Video Compression: H.264, JPEG, Motion JPEG
• Image Sensor: 1/2.5″, 5-Megapixel CMOS, 2,592 x 1,944 (pixels), exceptionally low noise levels and low-light sensitivity for superior video quality at very high resolution (1080p)
• Lens Type: 1/2″, 4.5-10mm, Manual-Iris, CS
• Day & Night: Electronic, with integrated IR_CUT and light sensor
• Responsivity: 1.4V/lux-sec (550nrn)
• Max Video Resolution: Full HD (16:9): 1920 x 1080p
• Pixel Dynamic Range: 70.1dB, with SNRMAX 38.1 dB (full resolution)
• Max Frame Rate in Max Resolution: 30fps
• Video Bit Rate: 32 Kbps — 8 Mbps
• Video Output: BNC, Voltage 1.0V PP, Resistance 75Ω
• Audio Input: 3.5mm LINE-IN, built-in Microphone
• Audio Output: 3.5mm LINE-OUT
• Alarm Input: Yes — One Interface, Normal Open
• Alarm Output: Yes — One Interface, 125VAC/0.5A or 30VAC/2A
• Audio Compression: G.711a/u
• Serial Ports: RS485
• IRIS Control: Manual IRIS (default lens), optional auto-iris lens
• Embedded Analytics: Motion detection (up to 16 target areas), video loss (pending)
• Pre-/post-alarm Buffer: 24MB
• Snapshots: Trigger upon events, send via email/FTP
• Multi-streaming-rate for Preview & Recording: Yes
• Security: Video watermark (pending), HTTPS, Password
• Network Port: 10M/100M auto-sensing, RJ45
• Network Protocol: TCP/UDP/IP, RTP/RTCP, RTSP, DHCP, PPPoE, DDNS, HTTP, HTTPS
• Power over Ethernet (PoE): Standard, IEEE 802.3af Class 0
• Peripheral Ports: SDHC 2.0, USB 2.0
• SIP/VoIP Support: Yes
• Dimensions (L x W x H): 152mm x 68mm x 64mm (Lens not included)
• Weight: 0.51kg (Lens not included)
• Temperature / Humidity: 0°C — 45°C (32°F-113°F); Humidity 10-90% RH (non—condensing)
• Power Adapter Output: 12VDC/1A; Input: 100-240VAC, 50-60Hz
• Regulatory Compliance: FCC Part 15, Subpart B Class B; CE EN 55022 Class B, EN 61000-3-2, EN 61000-3-3, N 55024, EN 60950-1; C-tick AS/NZS CISPR 22, CISPR 24

Recently, VoIPon had the opportunity to interview the founders of FreeSWITCH,  Anthony Minessale, Mike Jerris and Brian West. FreeSWITCH is billed as The World’s First Cross-Platform Scalable FREE Multi-Protocol Soft Switch and certainly recently, it’s one of the most talked about.

Below is a transcript of our call:

[VoIPon] I’m excited to learn more about what has been going on in the community and to bring VoIPon’s listener’s up to speed with what’s new at FreeSWITCH. Thanks for joining us today guys.

For those of us, like me, who have missed the last ClueCon, what are some of major achievements of FreeSWITCH in the last year or so?

[Anthony Minnesale] One of the biggest ones we announced during ClueCon was failure recovery. We added a feature to our SIP endpoint to allow it to use the core of FreeSWITCH to store state data about the call, which could be shared over to another machine or on the same machine; it’s used to resurrect the channels mid call and actually allow it to recover from a complete failure, such as like a machine hardware failure or software crashing, stuff like that.

And other than that it was mostly the new book that we wrote about FreeSWITCH 1.06 and we had that released shortly before the conference.

[VoIPon, Kathleen Reed] Anybody have anything else to add?

[AM] There is probably a thousand things, but to bring them all at the top at once…laughs.

[VoIPon] OK great.

[Mike Jerris] I think we’ve had some huge improvements as far as stability. Just, you know, as we’ve grown, just as we’ve started to add commercial products on top of it. We’ve focussed really hard and we’re just ratcheting up stability and a lot more things will just work automatically, as opposed to having lots of extra config options.

[VoIPon] Recently, Digium announced Asterisk SCF. What will this mean for FreeSWITCH?

[AM] I don’t really see there being much of an impact at this stage as it’s a new product. Really, it’s great to see them broadening their horizons. They’ve been focussing on just their hardware and Asterisk itself for a while so they’ve come up with a new idea, which is admirable. It has a pretty good chance of being a useful tool in the open source world, so we’ll see what they do.

FreeSWITCH has long been known for its robustness and scalability. Why should someone choose FreeSWITCH as a telephony platform for their organization NOW as opposed to Asterisk, and the new Asterisk SCF?

[AM] I don’t think there is a competition going on between Asterisk SCF and FreeSWITCH. Based on what I’ve learned from the announcement it’s used to join various locations by trying to provost a fault tolerance by redirecting calls when it notices something’s down and whatnot, which is actually, a compliment to software like FreeSWITCH.

They’ve already demonstrated that you would use Asterisk alongside of it as the termination point for the calls and whatnot and it would just be another tool that you could use.

There are several open source voice applications that are used together in a system, and we cooperate with all of the other open source projects. Most of our users are using all of these different things together, so I’m sure it won’t be any different.

[VoIPon] Is there anything else either of you would like to add, Brian, Mike?

[MJ] I think in general it’s a right tool for the job kind of problem to solve. FreeSWITCH and Asterisk takes different approaches to things, and tries to solve different problems. So like Tony said, they can complement each other well—for different sorts of problems.

[Brian West] I do think a lot of people do use FreeSWITCH and Asterisk together, they are complementing each other in multiple installs that our people in the community have.

[VoIPon] Businesses often times see a large community/user base as a sign of stability in an open source project, as in a lot of people are using it, therefore bugs will be found quickly and it’s more safe to use. How are you growing the FreeSWITCH community? What can VoIPon’s listeners do to help?

[AM] Well, I think the best way to grow the community is to just be inviting and convince the people who stick around to help foster in the new people.

We started with a community of four people. Within a year it had grown to about 60. Now we have 1000s of users on our mailing list and we probably don’t even have a way to measure everyone that is using FreeSWITCH. But we have a very active mailing list and we have a live chat in IRC and we sort of use that as an exit poll type thing. In an election, you don’t sample everybody, you take a sample of few things and you can kind of use it to gage the bigger numbers. We’ve watched the ratio of people in the chat room going up, assuming for every person who is in there, there is probably some large number of several hundred people to go along with it.  We’ve watched that number go up.

We’ve gone from like 5 people in our chat room to well over 230 on a daily basis. We’ve actually surpassed or been equal to the number of the people on the Asterisk channel at some times in the last couple of months.

Asterisk probably has a larger community because they have had five years head start on us. We were part of it originally and we learned a lot about voice from there and we started our own. I always sort of took a new attitude toward it where people would ask to come in and we’d say, come on in. People would ask for help, we’d say, join our chat room and we’ll help you. We help people out in the open. And I think, just being inviting and trying to help people when we can, and try to get other people to help, so if we help someone, they in turn help the next guy that comes along with the same problem, or help us document. That really helps spread out the huge amount of challenges we have. There is like endless amounts of stuff to do on a project like this, so the more the merrier.

[VoIPon] Wonderful, Mike, Brian, anything else to add on the community side of things?

[MJ] Sure, there’s two major aspects of community. There is the end user community and the developer community. And for the end user community, the biggest thing has always been trying to help people as much as you can, and treat people with respect. That’s been a big help in engaging people. The book has helped with that, as far as engaging people who aren’t necessarily online that we are interacting with daily.

On the developer’s side, I think we’ve been really proactive, when we find someone who is interested, and we find they have the skill set, that we’ve worked very closely with them. Because FreeSWITCH is so modular, we’ve had people adopt modules or many people have written their own modules, and we’ve given people easy access to be able to commit directly into the tree when they own something. To engage them more, share in ideas, just actively engaging people.

As far as how can new people help? If you’re a user, by all means, download it; try it out; give us feedback on what’s working for you and what’s not. For developers, if you have an interesting problem to solve, come on by, talk to us, and we’ll help steer you in the right direction.

[VoIPon] Brian, I thought I heard you start to say something there?

[BW] Basically, they both covered everything I would have said. Mike caught everything I was going to go over.

[VoIPon] Wonderful. And finally, what is the coolest feature of FreeSWITCH in your opinion and how does this relate to the future of the telephony space overall or does it?

[BW] I think one of the coolest features that was added recently was the deep machine stuff that Tony added for DTMF parsing. You can map digits to like transfer or other applications while you’re still in a phone call. For example, I can press *500 and it would play a file. Or, I could press *500 and it would do something else. I think that is one of the neatest things that’s been added. I don’t think that really matters in the future of telephony space. It might. But I think it’s a neat one.

[VoIPon] Excellent. Any other favourites you’d want to add, guys, or should we just wrap it up at that?

[MJ] My favourite feature is a little less specific. It is that FreeSWITCH offers you a toolbox for communications as a whole for easily building whatever you can imagine. So, instead of specific feature that you have so many possibilities literally. You can build things to make it very accessible to any way to interact with it. That as an overall feature you can build whatever you want, and we’ve given you the tools to do so.

[AM] The newest feature is always the favourite, because every time we make something else. The last big one that I added was to use the FSK Caller ID signal that is typically sent through an analogue phone to go across a VoIP call presumably going down and possibly across the PSTN. So if one FreeSWITCH machine encounters another one, with any amount of other stuff in the middle, it sends a small blurb through the audio path, that will actually tell them that they can be reached over the internet instead of through the PSTN and they can redirect the call on the fly right to the internet.

I’d call it an alpha feature at this point, because I just put it in the code tree and haven’t documented it yet.

[VoIPon] Excellent

[AM] It’s something we’ve talked about doing for a long time.

Anthony Minessale, Mike Jerris and Brian West are the founders of FreeSWITCH. For more information about FreeSWITCH please visit www.freeswitch.org or the annual telephony developer’s conference, ClueCon please visit www.cluecon.com.

Grandstream GXV3140 Illustration	Grandstream GXV3140 Illustration

Grandstream Networks, the leading manufacturer of IP voice/video telephony and video surveillance solutions, announced today its newest member of the GXV31xx series VoIP multimedia phones, the GXV3175 desktop IP multimedia telephone with a 7″ touch screen LCD. This is the sibling to the award-winning Skype-certified GXV3140, which has been highly popular amongst enterprise and SMB customers.

The sleek tablet-like GXV3175 delivers extraordinary video telephony capabilities with amazing picture sharpness, superior HD audio quality, and a variety of business productivity, casual entertainment, and social networking applications that are embedded in the phone. The GXV3175 represents the future of personal multimedia communications and redefines desktop IP communications experience with a raised level of innovation and integration of standard based HD IP telephony, high quality real time video conference, online rich media infotainment service, social multimedia and business productivity applications.

“The GXV3175 is an impressive value as it supports HD Voice, multiple SIP accounts, high quality video conferencing with TV output, WiFi, a Webkit-based Web browser, widgets and cloud-based photo album/social network connectivity,” said Rich Tehrani, CEO and group editor-in-chief of Technology Marketing Corporation (TMC). “The 7″ color touch screen coupled with a single home button makes the phone easy-to-use – like a smartphone, in fact.” He added, “In terms of bang for the buck, Grandstream has shown once again it can deliver a good looking phone with lots of features at a reasonable price level.”

Featuring 3-lines with up to 3 SIP accounts plus a FREE peer-to-peer IPVideoTalk service out of the box, the GXV3175 also includes:

High Quality Video and HD Audio: a 7″ touch screen color LCD, 1.3 mega pixel tiltable CMOS camera with privacy shutter, dual 10M/100M Ethernet ports with integrated PoE, integrated WiFi (802.11b/g/n), dual USB ports, SD port, stereo audio and TV output, headset jack, and stylus pen.

• Video Conferencing Projection: allowing connection to a digital TV for larger caller projection.
• Easy Touch-and-Swipe User Interface with Built-In Infotainment and Social Networking Applications: the phone offers an intriguing touch-and-swipe graphic user interface in multiple languages, a full Web browser, personalized local weather, news, stock, currencies, games, thousands of online radio channels, support for video and music streaming service such as YouTube and Last.fm, home automation and monitoring, slide show, digital photo frame, Web photo albums such as Yahoo Flickr/Photobucket/Phanfare, social networking applications such as Facebook and Twitter (pending), IM with Yahoo/MSN/Google (pending), and many more.
• Advanced Business Applications: the phone has a number of business productivity tools such as calendar, visual phone book directory, visual BLF extension and fast-dial keys, visual ring tone, mechanism for streaming multimedia advertisements, and HTML/JavaScript based SDK/API for many more advanced custom application development.
• Broad SIP Interoperability with Security Protection and Auto-Provisioning: the phone is interoperable with most 3rd party SIP devices, major IPPBX or softswitch platforms, supports strong security protection using HTTPS/TLS/SRTP/OpenVPN, and can be auto-provisioned using TR-069 standard for large volume deployment in enterprise or service providers networks.

“The Internet has become an integral part of many businesses activities and people’s lives,” said David Li, CEO of Grandstream. “Besides mobile phones, there is an emerging demand for an easy-to-use, always-on desktop communication device that provides instant access to people’s growing online communication needs for their business and personal lives without a PC.” He added, “We are very excited to announce the GXV3175 touch screen IP multimedia phone that offers a stunning experience of high impact visual and versatile communications for the enterprise and home market.”

VP-2009P represents the next generation in personal IP multimedia communication. Based on HD quality and other main functions of former phones, VP-2009P enabled an excellent face-to-face communication to provide a brand-new mode for business and conferencing. The extraordinary video quality, advanced telephony features, rich applications, fashionable outlook, sturdy and durable structure ensure VP-2009P as an ideal communication device for corporate office and residential users.

The comment from Broadsoft shows that Yealink manufactures very competitive IP phones with high cost performance and affordable price that are being used by Broadsoft’s customers including KPN, Telefonica, PCCW, Bezeq and SingTel. Besides supporting all of the enhanced BroadWorks features, Yealink phone is also integrated with BroadWorks device management. The Interoperability of VP-2009P this time creates an excellent value and innovation solution for both end-users and carriers and strengthens faith of Yealink in continuing customer-centric approach as well as friendly cooperation with Broadsoft.

About Broadsoft
Broadsoft provides software that enables fixed-line, mobile and cable service providers to deliver voice and multimedia services over their IP-based networks. The Company’s software, BroadWorks, enables service providers to provide enterprises and consumers with a range of cloud-based, or hosted, IP multimedia communications, such as hosted IP private branch exchanges, video calling, unified communications, collaboration and converged mobile and fixed-line services.

About Yealink
Yealink is professional designer and manufacturer of IP phones and video phones for the world-wide broadband telephony market. Yealink products are fully compatible with the SIP industry standard, and have broad interoperability with the major IP-PBX, softswitch and IMS on the market today. For more information please visit the Yealink Phone section.

“The D500 is very useful for companies and network operators that host IP PBX, call centers and VoIP services, where there is a great need for high port density,” said Doug Vilim, Vice President at Sangoma and a leading speaker at Elastix World. “The product is receiving a tremendous reaction from the market and we’re already getting a very high level of interest from customers all over the world.”

The D500 is both standalone and accessible via IP networks. This means it can be installed in the same server where your telecom applications run or it can be pooled in dedicated transcoding server farms for maximum deployment flexibility. Either way, the voice transcoding processing is performed on the card, freeing up the host CPU to do other tasks. Most IP telephony applications require the use of multiple types of voice codecs, which are used to digitally compress the voice signals. This can be done to save bandwidth by using G.729 codecs or to enhance the voice signal by using HD codecs (such as G.722.1). As more and more VoIP networks get deployed and interconnected, there are instances where there will be mismatches of codec capabilities between two endpoints, creating the need to transcode (mediate and adapt between various codecs).

The D500 card, available in a compact PCIe form factor, offers developers and integrators the ability to convert up to 2,000 sessions on a single board and supports a long list of codecs such as G.711, G.723.1, G.729ab, HD Voice (G.722.1) GSM and AMR. Moreover, these codecs are generally indemnified, meaning there are no additional costs other than the price of the D500 hardware.

“The D500 card from Sangoma is really unique in that it opens up a whole new set of opportunities for application developers using computer telephony as their platform for development,” said Regis du Manoir, VP of business development at Dip International, a company that represents manufacturers that specialize in highly integrated semiconductors and board solutions for major markets such as communications, VoIP and Video over IP. “With this kind of port density, and with the headaches of codecs licensing removed, it makes it much easier for integrators to address large scale deployments.”

The new transcoding portfolio from Sangoma is used on both open source and proprietary applications. In the open source world, the D500 software drivers (like those of Sangoma’s other transcoding cards), provide “plug-and-play” capabilities for both Asterisk® and FreeSWITCH®, two leading open source telephony projects. With the compatible drivers, the open source telephony platforms can use the D500 cards as seamless voice transcoding resources. This means that existing Asterisk and FreeSWITCH applications can readily start leveraging the D500 capabilities. Further, the open source telephony software can be used as a gateway or session border controller to provide network-based transcoding services.