The new butterfly handset is designed especially for carpeted environments, and offers the most important features and functionalities within basic voice and messaging.

The new KIRK Butterfly handset will have a stylish and contemporary design, and a TFT color display that provides a clear view. It will be available in a range of fashionable colors, which enables you to choose the color that best suits your organisation.

Furthermore, a range of accessories will be created for the handset in order for you to fully embrace the possibilities of being wireless.

For more information and product details please view the Polycom Kirk DECT Butterfly Handset here

Grandstream’s GXP1400 IP Phone and GXP1405 IP Phone offers cutting edge VoIP telephony features to small to medium-sized businesses (SMBs) with or without an IP PBX, at a revolutionary price-point. Phone users have access to an extensive set of best-in-class audio, visual, security, provisioning, and business features that allow the GXP1400/1405 to create a class of its own in the affordability segment of the IP phone market. The GXP1400/1405 features built-in TLS/SRTP security, secondary SIP server, best-inclass HD audio including an acoustically redesigned HD handset, improved double-talk performance, Acoustical Echo Cancellation on its full-duplex HD speakerphone, easy automated provisioning with TR-
069 or HTTPS, advanced XML customization and more.

The GXV3500 is a first-of-its-kind security, surveillance and facility management solution for SMBs that want to preserve their investment in traditional CCTV analog cameras while adding the capability to remotely display live video on a TV, or use the device as a voice/video public announcement system (PAS) over an IP network. The GXV3500 IP Video Encoder+Decoder+PAS is a 3-in-1 combo device that combines advanced single channel H.264 real time video compression or decompression functions at up to D1 resolution and up to 30fps in a single box. It also features two-way audio paging and video streaming using SIP/VoIP, integrated PoE (Power over Ethernet), serial port for PTZ camera control, microphone input, audio/video output, alarm inputs & outputs, pre- and post-event recording buffer, as well as advanced encryption for security protection.

The GXV3500 can also uniquely function as both a voice and video PAS for office floors, lobbies, shopping malls, warehouses, waiting rooms, supermarkets, public transportation stations, etc. Users can speak from an attached external microphone, or call into the device from a SIP IP phone to make voice announcements or supply digital background music to public areas through an external loudspeaker (with integrated power amplifier), or call into the device from a SIP video phone to make live video announcement or video advertisement on a TV connected with the GXV3500 from anywhere in the world.

“I am happy to grant Grandstream with two Product of the Year Awards. The editors of INTERNET TELEPHONY have verified that the GXP1400 HD IP phone and GXV3500 IP Video
Encoder/Decoder/PAS display quality and innovation plus satisfy real needs in the marketplace,” stated Rich Tehrani, CEO, TMC. “I would like to congratulate the entire team at Grandstream for their
commitment to advancing IP communication technologies.”

ITEXPO EAST, MIAMI, FL Booth 517 (February 1, 2012) – Grandstream Networks, the leading manufacturer of IP voice/video telephony and video surveillance solutions, today announced the release of the HT701 a new 1-port Analog Telephone Adapter (ATA). The HT701 Analog Adapter is designed for single line analog phone users – residential, small-to-medium-sized business (SMB) and telecommuters – needing a compact, inexpensive ATA that converts legacy analog telephones, fax machines, conference telephones and other analog devices to access Internet-based telephone services. ATAs allow businesses and consumers to extend the value of their investment in analog devices while migrating to richer, less costly, and multimedia IP services at their own pace.

The HT701 is the newest addition to Grandstream’s HandyTone ATA/IAD series of VoIP access devices that offer a range of SIP standard-based ATAs for mixed VoIP applications. Its feature highlights include:

• Ultra compact design (.98in H x 2.6in W x 3.38in L) and extremely affordable cost
• Support 5 REN over up to 1km on 24 AWG line
• Advanced telephony features including Caller ID for various countries, call waiting, do-no-disturb,
  3-way conference, transfer, forward, message waiting, T.38 fax, flexible dial plan, multi-language
  voice prompt, etc.
• Comprehensive voice codecs including G.711 with Annex I/II, G.729A/B, G.723.1, iLBC, G.726
• Strong security protection of voice/data privacy using TLS/SRTP/HTTPS, secure and automated provisioning using TR069, HTTP/HTTPS/TFTP and AES encryption

Currently celebrating its 10-year anniversary in the VoIP marketplace, Grandstream was founded in 2002 selling initially an ATA and IP phone model. Since then, Grandstream has rapidly grown to be a globally recognized provider of a comprehensive portfolio of VoIP communications devices including IP phones, gateways, ATAs, video phones, and video cameras.

“We are excited to announce the new HT701 ATA for the residential Internet telephony service provider and small business market,” said David Li, CEO of Grandstream Networks. “The HT701 is the culmination of our 10 years’ experience developing award-winning advanced VoIP products. Its ultra-compact design, rich feature set, and unbeatable price point will make it a very popular product in the marketplace quickly.”

The Grandstream HT701 will be on display this week at Grandstream’s booth at ITEXPO East Miami, Booth 517.

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.

Commenting on the alliance, Michael Bibelman, Voiceserve’s CEO, said, “Combining Voiceserve’s cutting-edge VoIP software platform with Grandstream’s advanced IP equipment into one marketing promotion is a win-win for both companies.  The combined worldwide reach of both companies, together with each company’s well recognized products in this fast growing VoIP market should prove to be a dynamic marketing alliance.”

“We are pleased to  announce the interoperability between Grandstream’s IP voice & video product lines and Voiceserve’s VoipSwitch platform”, said  David Li, CEO of Grandstream Networks.  “This mutual certification ensures our common customers  will have  a plug-and-play and end-to-end solution with very competitive performance and economic attraction.”

The new GXP1100/GXP1105 target those customers who want an entry-level, flexible, and high performance phone with HD audio and robust telephony functionalities but don’t need to invest in more sophisticated multi-line phone models with LCD displays. The GXP1100/1105 supports a number of popular telephony features of Grandstream’s more advanced GXP21xx/14xx series HD phones,  such as security protection using TLS/SRTP/HTTPS, 802.1x, automated provisioning using AES encrypted XML configuration file and TR069, comprehensive voice codecs including the wideband G.722, etc.

More specifically, the two new models of GXP phones support these features:

GXP1100: entry level business HD IP telephone supporting single SIP account, up to 2 calls, 4 programmable keys, HD Handset with wideband audio, single 10/100Mbps network port, wall-mountable design with a desktop stand, 7 dedicated function keys for Hold, Flash/Call-Waiting, Transfer, Message, Mute, Volume, Dial/Redial.  The phone is energy green efficient with an average power consumption of less than 1W. Its Web configuration interface also supports multi-languages including English, German, French, Spanish, Italian, Russian, Chinese, Korean, and Japanese.

GXP1105: Includes all of the same features and functionalities of the GXP1100 but it additionally offers integrated Power-over-Ethernet (PoE).

“In 2011 Grandstream placed focus on, and successfully gained significant momentum in, the sub-$99 IP phone category by introducing a wide assortment of HD IP phones that meet the varying size and application needs of SMBs and enterprise customers,” said David Li, CEO of Grandstream Networks. “The release of the new GXP 1100/1105 phones demonstrates our continued commitment to serving this fastest growing market segment with a variety of innovative, high quality and affordable product offering.”

For more information on the GXP1100/1105 please Contact Us.

The vulnerability affects all Yealink devices, including those NOT purchased directly from VoIPon.

Some revisions of Yealink firmware will also allow an attacker to gain complete control of your IP Phone and SIP extension details.

Vulnerability information

Yealink phones are factory set with default usernames of “admin” and “user”, along with default passwords. Attackers can use several utilities to easily exploit this and instruct the phone to make calls without your permission, commonly to high cost international destinations.

Yealink are currently working on an update to their firmware to resolve this problem, in the meantime the information below should be followed for all Yealink handsets deployed within your organisation.

Recommendations

It is important that you follow the below steps to reduce your risk of fraudulent activity.

If you have purchased your phone from VoIPon configured to an extension either via our online store or from our sales team, then we are taking steps to ensure that your firmware is upgraded automatically and that the “user” password is changed. You should also ensure that the “admin” password is changed to something more secure.

If you have not purchased your Yealink phone from VoIPon or have made technical changes to how the device provisions then please consult with your supplier or IT department for any additional information that may be required.

Ensure that your phone is running the latest release of firmware from the list below. If it is not then please update your device immediately by clicking on the relevant link below. This will download the latest firmware, which you can then upload to your phone.

Yealink T20P 9.60.23.14: Firmware
Yealink T22P 7.60.23.14: Firmware
Yealink T26P 6.60.23.14: Firmware
Yealink T28P 2.60.23.14: Firmware

If you are unsure of the firmware upgrade procedure, a guide can be found on the Yealink FAQ.

Ensure that both the “admin” and “user” logins for the phones web interface are protected with a strong password. We recommend using this free online password generator to create a suitable password, which can then be updated using the phone’s web based administration.

• Save money through Least Cost Routing
• Turn Mobile phones into Office Phones
• Receive & send SMS per email & vise versa
• Provide Security: Failover routing to GSM
• Turn your beroFix into a VoIP to GSM Gateway

The bf2GSM module is a 2 port GSM module for the berofix Mainboards and Boxes.

• The BF2GSM is used for small enterprises who need up to 2 concurrent GSM calls. It can be used for mobile convergence, LCR and SMS.
• Our GSM Module is fully compatible and combinable with all berofix Modules and Cards.
• Check back later or sign up for our mailing list to be the first when it comes out.

Buy the card that fits your needs

Our easy-to-use configurator helps you select the right card and modules to match your demands. Click the configure button on the right to start.

• General Info
• Specifications
• Modules & Accessories

The bf2GSM module is a 2 port GSM module for the berofix Mainboards and Boxes.

The BF2GSM is used for small enterprises who need up to 2 concurrent GSM calls. It can be used for mobile convergence, LCR and SMS. Our GSM Module is fully compatible and combinable with all berofix Modules and Cards.

Specifications:

• 2 separate GSM Ports
• QuadBand(850/900/1800/1900 Mhz)
• GSM/GRPS Protocol Stack 3GPP Release 4 compliant
• SMS (Text and PDU)
• Integrated SMS-Mail and Mail-SMS Gateway
• GSM Supplementary Services (CLIP, CLIR)
• Dimension: 105 x 46 x 9 mm
• Weight: < 40g
• Operating Temperature: 0-70°C
• CE compliance

Modules

• 1 Port PRI Module (BF1E1)
• 2 Port PRI Module (BF2E1)
• 4 Port BRI Module (BF4S0)
• 2/2 Port BRI/FXS Hybrid Module (BF2S02FXS)
• 4 Port FXO Module (BF4FXO)
• 4 Port FXS Module (BF4FXS)
• 2 Port GSM Module (BF2GSM coming soon)

Accessories

• BFBridge (to use all 4 RJ45 slots on a Baseboard with one bf4S0, bf4FXS, bf4FXO, bf2S02FXS)
• BFTAdapters (to connect 2 BRI or analog Lines on one RJ45 of the berofix)
• BFPCM (PCM Cable to interconnect berofix Baseboards)
• BFE1Cross, (1.8m E1/T1 “cross-over” cable)
• BF19bracket (19 inch rack mount bracket for berofix Boxes)
• BFANTS (small 7 cm stock antenna for BF2GSM modules)
• BFANTL (2m cable antenna with magnetic foot for BF2GSM modules)
• BFFXXCable (RJ45 to RJ11 cable to use with berofix analog modules)

Add the GSM module to your beroFix installations and …

• Save money through Least Cost Routing
• Turn Mobile phones into Office Phones
• Receive & send SMS per email & vise versa
• Provide Security: Failover routing to GSM
• Turn your beroFix into a VoIP to GSM Gateway

!COMING OCTOBER 2011!

• Support for ISDN BRI point to point and point to multipoint lines
• The availability of configuration templates, allowing direct NBE configuration integrated with I-PBX platforms
• Smart updates. You can now update and upgrade the software directly from the Web management interface, without losing your configuration
• Compatibility with 3CX and SnomOne IPBX systems

Specification

We would also like to remind our customers that NBE is free to download and use for up to 12 ports. If you need more capacity, you need to purchase a software license. All you need to get started is a Sangoma compatible card. Not sure which cards are compatible? Please Contact Us for more information.

NetBorder Express software, together with Sangoma’s computer telephony boards, is a complete SIP-compliant VoIP Media Gateway solution, delivering the most flexible and cost effective solution by removing the need for external gateways.

Compatible with Analog FXO and FXS, MFC/R2, ISDN BRI and PRI telephony interfaces, NetBorder Express can scale from 2 to 480 simultaneous calls, providing excellent value and flexibility for deployments in any geography.

NetBorder Express communicates with applications such as IP-PBXs, speech enabled IVRs, conferencing servers, contact center or trunking gateways directly with the ubiquitous SIP control protocol to reduce deployment time, complexity and cost for next generation network application deployments.

NetBorder Express software is licensed via a simple online process. You can download a free SOHO version, which is full-featured for up to 12 ports. If you need more than 12 ports, a commercial license is required. Sangoma Telephony hardware must be purchased separately.

NetBorder Express is a full-fledged VoIP Gateway delivered as a software application. NetBorder Express version 4.0 support the following:

PSTN protocol termination:

• Analog FXO/FXS (with A200 or B600 hardware)
• ISDN PRI (with A100 hardware)
  - North American variants
  - EuroISDN
  - NEW Japan variant (INS1500)
• NEW ISDN BRI (with A500 and B700 hardware)
• NEW MFC/R2 (with A100 hardware)
• Capacity from 2 to 480 channels of simultaneous calls per system

VoIP protocol termination:

• TCP or UDP
• SIP Control Protocol per RFC 3261
• RTP/RTCP per RFC 3550/3551

Security

• TLS per RFC 5246
• SRTP per RFC 3711
• NEW HTTP Digest authentication

Media Processing

• G.168-2002 Echo Cancellation with hardware option
• DTMF over RTP per RFC 2833. International and custom tone detection and generation
• Voice Activity Detection and Comfort Noise Generation (VAD/CNG)
• NEW T.38 Fax relay

Software License

• NEW Default SOHO license available for download for up to 12 ports
• Commercial software licenses required above 12 ports
  - 25 digits license ID provided upon purchase
  - License ID and Server MAC address required for Web registration and download of software
  - License file host locked

Application Compatibility:

• Microsoft Lync
• 3CX
• SnomOne
• Voxeo
• Genesys SIP Server
• Any other standard SIP application
• NetBorder Express can be installed on the same server or on a dedicated server

Compatible Sangoma Hardware:

• Analog FXO and FXS
  - A200
  - NEW B600
• ISDN BRI
  - NEW A500
  - NEW B700
• MFC/R2
  - A101, A102, A104, A108
• ISDN PRI
  - A101, A102, A104, A108
• Notes:
  - Hardware must be purchased separately
  - Hardware echo cancellation option is mandatory for NetBorder ExpressNetBorder Express software installs compatible board drivers automatically. There is no need to obtain board drivers separately
  - Each specific NetBorder Express software version is tested against a specific version of the firmware that comes loaded on Sangoma hardware. It may be required to upgrade the firmware of the boards at installation time. At installation time, NetBorder notifies the administrator and provides instructions to perform this step
  - Consult NetBorder Express Release Notes for more details

Management and Configuration

• Web Server
• Web Services API
• Configuration files for easy integration into OEM applications
• SNMP DS0, DS1 and Call Stats MIBS
• Contextual Help Menus
• Extensive Logging
• Automatic Board Discovery
• Open Protocol Translation and XML based routing rules
• Configurable ISDN PRI to SIP Mapping per RFC 3398

Minimum Server Specs

• Intel or AMD processors
• Dual Core processor or later
• 1 GB of RAM or better