HARDWARE SOLUTIONS
VOIP GATEWAYS
Iristel’s range of VoIP Gateways and IP Phones are ideal for both residential and enterprise customers.
DIGITAL IP PHONES
Grandstream DP715 DECT
$109.99
Grandstream GXP1628
0
Grandstream GXP2140
0
Grandstream GXV3275
0
DP715/710 is the next generation of powerful, affordable, high quality and simple to configure VoIP DECT phones for small business and residential users. Their compact size, superb voice quality, rich feature set, market leading price-performance and wide range radio coverage enable consumers to maximize the power of IP voice application and mobility for a minimum investment. DP715/710 is SIP and DECT compliant and field proven for flexible deployment.
The GXP1628 is a powerful Gigabit IP phone designed for small businesses. This Linux-based, 2-line IP Phone model includes 8 BLF keys and 3-way conferencing to keep workers in-touch and productive. A 132x48 backlit LCD screen creates a clear display for easy viewing. Additional features such as dual HD audio, multi-language support, integrated PoE and 3 XML programmable allow the GXP1628 to be a high quality, versatile and dependable office phone.
A versatile Enterprise IP phone, the GXP2140 supports 4 lines, includes Gigabit ports and is compatible with Grandstream’s GXP2200 LCD extension module (GXP2200 EXT) making it ideal for receptionists and users who handle high call volumes. The GXP2140 includes HD audio, a 4.3 inch color screen, 5-way voice conferencing, and built-in Bluetooth support allows for use of Bluetooth headsets and transferring of phonebook and calls from paired mobile devices, the GXP2140 is perfect for Enterprise & SMB customers with the need for quality and versatility in their desktop communications.
The GXV3275 Video IP Phone for Android combines a 6-line IP video phone with a multi-platform video conferencing solution and the functionality of an Android tablet to offer an all-in-one communications solution. The GXV3275 runs the Android Operating System and therefore offers full access to the many Android apps in the Google Play Store, including popular productivity and business apps.
Panasonic KX-TGP600
0
Panasonic KX-TPA60
0
Panasonic KX-TPA65
0
Polycom 335
$209.99
The Panasonic TGP600 Cordless DECT Phone with Repeater is a great DECT phone solution that can sustain up to 8 simultaneous network conversations, handsets, and also SIP registrations.
This entry level model features a base station unit that can be wall mounted, and also comes with a top of the line SIP cordless handset with charging station. Containing high-quality wideband voice services, as well as noise reduction, this unit is equipped with a phone book of up to 500 contacts and nearly 200 hours of stanby and 11 hours of talk time. The TGP600 also has a repeater that registers up to 6 lines. With a 1.8 inch, 65,000 color TFT display, this cordless handset and base station unit will meet the needs of your business.
• VoIP Support: IETF SIP Version 2, Broadworks/Broadsoft Inc., Asterisk/Digium Inc.
• Voice Codec: G.722/G.722.2 (AMR-WB), Narrowband G.711a-law/G.711u-law/G.729a
• Network: (1) 10/1000 base-T auto MDI/MDIX Ethernet LAN port, IP Stack mode of IPv4, IPv6, IPv4/IPv6 Dual
• Security: Secure RTP, SIP/SIP-TLS
• Provisioning: MTTP/MTTPS/FTP/TFTP and Local/Remote WEB Config.; Disabling provisioning feature of local operation (yes)
• QoS Support: DSCP, IEEE802.1q, TOS
• Key and Indications: LED Status Indicator (Base Unit), LCD Handset 1.8 in color screen
This entry level model features a base station unit that can be wall mounted, and also comes with a top of the line SIP cordless handset with charging station. Containing high-quality wideband voice services, as well as noise reduction, this unit is equipped with a phone book of up to 500 contacts and nearly 200 hours of stanby and 11 hours of talk time. The TGP600 also has a repeater that registers up to 6 lines. With a 1.8 inch, 65,000 color TFT display, this cordless handset and base station unit will meet the needs of your business.
• VoIP Support: IETF SIP Version 2, Broadworks/Broadsoft Inc., Asterisk/Digium Inc.
• Voice Codec: G.722/G.722.2 (AMR-WB), Narrowband G.711a-law/G.711u-law/G.729a
• Network: (1) 10/1000 base-T auto MDI/MDIX Ethernet LAN port, IP Stack mode of IPv4, IPv6, IPv4/IPv6 Dual
• Security: Secure RTP, SIP/SIP-TLS
• Provisioning: MTTP/MTTPS/FTP/TFTP and Local/Remote WEB Config.; Disabling provisioning feature of local operation (yes)
• QoS Support: DSCP, IEEE802.1q, TOS
• Key and Indications: LED Status Indicator (Base Unit), LCD Handset 1.8 in color screen
Compatible with the KX-TGP600 smart IP wireless phone system, the KX-TPA60 wireless (DECT) handset allows businesses to expand their communications as they grow. When combined with the KX-A406 repeater, they provide wide-ranging coverage that integrates flexibility and enhanced customer service throughout the operations of a business.
Compatible with the KX-TGP600 smart IP wireless phone system, the KX-TPA65 wireless desk phone offers a significant advantage over ‘traditional’ desk phones. While it features the same range of desktop functionality as its wired competitors, the phone’s DECT capabilities means no wired LAN is required when installation is being carried out, making the process more convenient and straightforward.
The Polycom SoundPoint IP 335 phone delivers a business grade telephony endpoint at an entry-level price.
The Polycom SoundPoint IP 335 features Polycom HD Voice, bringing life-like richness and clarity to every call. Polycom HD Voice incorporates wideband audio for over twice the voice clarity. Like other Polycom Phones with HD Voice, the Polycom IP SoundPoint 335 features Polycoms patented Polycom Acoustic Clarity technology for crystal-clear, noise and echo-free sound; and best-in-class system design for high-fidelity voice reproduction.
The Polycom SoundPoint IP 335 desktop phone also delivers advanced features and capabilities such as call hold, pick-up, transfer, and three-way local conferencing ,as well as more advanced capabilities such as shared call/bridged line appearance, built-in XML microbrowser, and RJ-9 headset port.
The Polycom SoundPoint IP 335 phone is designed to bring a high-quality, cost effective solution to cubicle workers/call center operators through advanced telephony features and HD Voice technology, making voice communications more effective and productive.
The Polycom SoundPoint IP 335 features Polycom HD Voice, bringing life-like richness and clarity to every call. Polycom HD Voice incorporates wideband audio for over twice the voice clarity. Like other Polycom Phones with HD Voice, the Polycom IP SoundPoint 335 features Polycoms patented Polycom Acoustic Clarity technology for crystal-clear, noise and echo-free sound; and best-in-class system design for high-fidelity voice reproduction.
The Polycom SoundPoint IP 335 desktop phone also delivers advanced features and capabilities such as call hold, pick-up, transfer, and three-way local conferencing ,as well as more advanced capabilities such as shared call/bridged line appearance, built-in XML microbrowser, and RJ-9 headset port.
The Polycom SoundPoint IP 335 phone is designed to bring a high-quality, cost effective solution to cubicle workers/call center operators through advanced telephony features and HD Voice technology, making voice communications more effective and productive.
100 or more contacts
Voice Mail
3-way Call
Call Hold
Mute Key
Quick Redial
Caller ID Show/Restrict
Call Forward
Volume Keys
102 x 33 pixel Graphical LCD
2 Calling Lines
Polycom 450
$289.99
Polycom 550
$299.99
Polycom 650
$369.99
Polycom 670
$509.99
The Polycom SoundPoint IP 450 desktop phone is designed to bring advanced telephony features and applications to cubicle/office workers handling a moderate volume of calls. With its high-resolution, graphical backlit display supporting multiple languages and Asian characters, applications enabled XML Microbrowser, and Polycom HD Voice, the SoundPoint IP 450 sets new standards for the mid-range SIP desktop phone.
The SoundPoint IP 450 features Polycom HD Voice, bringing life-like richness and clarity to every call. Polycom HD Voice incorporates wideband audio for over twice the voice clarity, Polycom's patented Acoustic Clarity Technology for crystal-clear, noise and echo-free sound, plus best-in-class system design for high-fidelity voice reproduction.
An enterprise-grade phone, the SoundPoint IP 450 delivers an easy-to-navigate menu and a combination of 17 dedicated hard keys and four content sensitive soft keys on a backlit, multi-level grayscale 256 x 116 pixel LCD display. Hosting a built-in XML Microbrowser, the SoundPoint IP 450 offers an easy- to-access graphical interface to run productivity-enhancing third-party applications using Polycom's flexible and open XML API.
The SoundPoint IP 450 features Polycom HD Voice, bringing life-like richness and clarity to every call. Polycom HD Voice incorporates wideband audio for over twice the voice clarity, Polycom's patented Acoustic Clarity Technology for crystal-clear, noise and echo-free sound, plus best-in-class system design for high-fidelity voice reproduction.
An enterprise-grade phone, the SoundPoint IP 450 delivers an easy-to-navigate menu and a combination of 17 dedicated hard keys and four content sensitive soft keys on a backlit, multi-level grayscale 256 x 116 pixel LCD display. Hosting a built-in XML Microbrowser, the SoundPoint IP 450 offers an easy- to-access graphical interface to run productivity-enhancing third-party applications using Polycom's flexible and open XML API.
3 Calling Lines
100 or more contacts
Caller ID Show/Restrict
Call Hold
3-way Call
Call Forward
Mute Key
Volume Keys
Voice Mail
Quick Redial
LCD monochrome display (256 x 116 multi-layer)
The SoundPoint IP 550 desktop phone is a four-line SIP phone that delivers calls of unprecedented clarity and supports a comprehensive range of cutting-edge features.
The SoundPoint IP 550 desktop phone features Polycom's revolutionary HD Voice technology, which brings life-like richness and clarity to every call.
Polycom HD Voice technology incorporates wideband audio for over twice the voice clarity. Add the support of four lines, a backlit, high resolution, easy-to-read graphical display and flexible customizations options, and it becomes clear why the SoundPoint IP 550 is certain to meet the voice communication needs of the most demanding managers and professionals.
The SoundPoint IP 550 desktop phone features Polycom's revolutionary HD Voice technology, which brings life-like richness and clarity to every call.
Polycom HD Voice technology incorporates wideband audio for over twice the voice clarity. Add the support of four lines, a backlit, high resolution, easy-to-read graphical display and flexible customizations options, and it becomes clear why the SoundPoint IP 550 is certain to meet the voice communication needs of the most demanding managers and professionals.
LCD monochrome display (320 x 160)
Caller ID Show/Restrict
Quick Redial
Volume Keys
Mute Key
Call Hold
Call Forward
3-way Call
Voice Mail
100 or more contacts
4 Calling Lines
The SoundPoint IP 650 is the first IP phone to use Polycom's revolutionary HD Voice technology that delivers voice communications of life-like richness and clarity. In addition to the advancements in voice clarity Polycom has also advanced many of the Sounpoint 650's features and applications. The Soundpoint 650 phone's SIP 2.0 software fully supports Microsoft Live Communications Server 2005 for telephony and presence, and integrates with Microsoft Office Communicator instant messenger client. The SoundPoint IP 650 features a USB port for future applications.
The SoundPoint IP 650 accommodates 6 lines in a standalone mode and up to 12 lines when equipped with SoundPoint IP Expansion Modules, as an attendant console. The phone supports shared call / bridged line appearances as well as busy lamp field (BLF) functionality that enables phone attendants to monitor and manage calls more efficiently. When equipped with up to three SoundPoint IP Expansion Modules, the SoundPoint IP 650 delivers the advanced call handling capabilities and enhanced user interface of a high-performance attendant console.
The SoundPoint IP 650 delivers all of its capabilities through an intuitive user interface, featuring a high-quality backlit 320 x 160 graphical grayscale LCD display, an easy-to-navigate menu, and a combination of dedicated keys and context-sensitive soft keys for one-button access to essential telephony features.
The SoundPoint IP 650 accommodates 6 lines in a standalone mode and up to 12 lines when equipped with SoundPoint IP Expansion Modules, as an attendant console. The phone supports shared call / bridged line appearances as well as busy lamp field (BLF) functionality that enables phone attendants to monitor and manage calls more efficiently. When equipped with up to three SoundPoint IP Expansion Modules, the SoundPoint IP 650 delivers the advanced call handling capabilities and enhanced user interface of a high-performance attendant console.
The SoundPoint IP 650 delivers all of its capabilities through an intuitive user interface, featuring a high-quality backlit 320 x 160 graphical grayscale LCD display, an easy-to-navigate menu, and a combination of dedicated keys and context-sensitive soft keys for one-button access to essential telephony features.
100 or more contacts
3-way Call
Voice Mail
Call Forward
Call Hold
Mute Key
Volume Keys
Quick Redial
Caller ID Show/Restrict
LCD monochrome display (320 x 160)
6 Calling Lines
The SoundPoint IP 670 is an application-enabled desktop IP phone with a high-performance color display, Polycom's revolutionary HD Voice for unparalleled voice quality, and Gigabit Ethernet connectivity.
It is designed to provide professionals with a vibrant color interface for easier viewing and navigation, as well as a high level of integration with productivity-enhancing applications and business processes. The SoundPoint IP 670 integrates Polycom HD Voice to bring life-like richness and clarity to every call.
The SoundPoint IP 670 accommodates six lines in standalone mode. When equipped with up to three SoundPoint IP Color Expansion Modules, the SoundPoint IP 670 becomes a 34-line, productivity-enhancing attendant console for attendants to increase call handling capabilities and to view presence information without having to be in front of a PC.
It is designed to provide professionals with a vibrant color interface for easier viewing and navigation, as well as a high level of integration with productivity-enhancing applications and business processes. The SoundPoint IP 670 integrates Polycom HD Voice to bring life-like richness and clarity to every call.
The SoundPoint IP 670 accommodates six lines in standalone mode. When equipped with up to three SoundPoint IP Color Expansion Modules, the SoundPoint IP 670 becomes a 34-line, productivity-enhancing attendant console for attendants to increase call handling capabilities and to view presence information without having to be in front of a PC.
Backlit color display (320 x 160)
Caller ID Show/Restrict
Quick Redial
Volume Keys
Mute Key
Call Hold
Call Forward
3-way Call
Voice Mail
100 or more contacts
6 Calling Lines
Polycom VVX 1500 Video Telephone
$879.99
Polycom VVX 300/310
0
Polycom VVX 400/410
0
Polycom VVX 500
$339.99
The Polycom VVX 1500 dual stack business media phone unifies video, voice and applications capabilities in a simple-to-use personal communication solution. With its unique touch screen interface, the VVX 1500 makes video calls as simple as using a desktop phone. Its large display and ease of use make the VVX 1500 an ideal productivity tool for today's busy executives and professionals, whether they are in office, retail, professional services, or healthcare environments.
The Polycom VVX 1500 includes an integrated camera with multiple adjustable elements including camera tilt, base height and screen angle to suit the environment and a user's preferences. The VVX 1500 comes with a Web service called Polycom My Info Portal, through which customers can select to receive personalized Web content, such as stock prices, weather, and news. With native H.323 support, it connects easily with all standards-based H.323 video conferencing and telepresence systems. The VVX 1500 also features an open API and WebKit-based full browser that enable third-party developers to create applications that integrate the VVX 1500 with business systems such as UC, customer relationship management (CRM), and other vertical business applications.
The Polycom VVX 1500 is a business media phone for H.323 and SIP environments combining one-touch video calling, integrated business applications, and advanced IP telephony in a flexible, future-proof Unified Communications (UC) solution.
The Polycom VVX 1500 includes an integrated camera with multiple adjustable elements including camera tilt, base height and screen angle to suit the environment and a user's preferences. The VVX 1500 comes with a Web service called Polycom My Info Portal, through which customers can select to receive personalized Web content, such as stock prices, weather, and news. With native H.323 support, it connects easily with all standards-based H.323 video conferencing and telepresence systems. The VVX 1500 also features an open API and WebKit-based full browser that enable third-party developers to create applications that integrate the VVX 1500 with business systems such as UC, customer relationship management (CRM), and other vertical business applications.
The Polycom VVX 1500 is a business media phone for H.323 and SIP environments combining one-touch video calling, integrated business applications, and advanced IP telephony in a flexible, future-proof Unified Communications (UC) solution.
Caller ID Show/Restrict
Quick Redial
Volume Keys
Voice/Video Mail
6 Calling Lines
100 or more contacts
3-way Call
Call Forward
Call Hold
Mute Key
Ringtones
7" TFT LCD display (adjustable screen angle)
Web Browser
These powerful 6-line entry-level Business Media Phones are for today's cubicle workers that handle a low to moderate volume of calls and need crystal clear communications.
• Backlit grayscale graphical LCD (208 x 104)
• 6 line or speed dial keys
• HD Voice up to 7KHz on all audio paths (Speaker, Handset, Headset)
• 2 x Ethernet 10/100 or GigE (VVX 310)
• Asian character support
• Hard Keys: 12-key dial pad, home, speaker, mute, headset, volume, messages, hold, transfer
• 4-way navigation cluster with center "select" key
• Supports VVX Expansion Module and VVX Color Expansion Module (Expandability up to 3 modules)
• Backlit grayscale graphical LCD (208 x 104)
• 6 line or speed dial keys
• HD Voice up to 7KHz on all audio paths (Speaker, Handset, Headset)
• 2 x Ethernet 10/100 or GigE (VVX 310)
• Asian character support
• Hard Keys: 12-key dial pad, home, speaker, mute, headset, volume, messages, hold, transfer
• 4-way navigation cluster with center "select" key
• Supports VVX Expansion Module and VVX Color Expansion Module (Expandability up to 3 modules)
These color 12-line mid-range Business Media Phones are for today's office workers and call attendants who depend on crystal clear communications
• 3.5" TFT (320 x 240)
• 12 lines or speed dial keys
• HD Voice up to 7KHz on all audio paths (Speaker, Handset, Headset)
• 2 x Ethernet 10/100 or GigE (VVX410 )
• Asian character support
• Hard Keys: 12-key dial pad, home, speaker, mute, headset, volume, messages, hold, transfer
• 4-way navigation cluster with center “select” key
• Supports VVX Expansion Module and VVX Color Expansion Module (Expandability up to 3 modules)
• 3.5" TFT (320 x 240)
• 12 lines or speed dial keys
• HD Voice up to 7KHz on all audio paths (Speaker, Handset, Headset)
• 2 x Ethernet 10/100 or GigE (VVX410 )
• Asian character support
• Hard Keys: 12-key dial pad, home, speaker, mute, headset, volume, messages, hold, transfer
• 4-way navigation cluster with center “select” key
• Supports VVX Expansion Module and VVX Color Expansion Module (Expandability up to 3 modules)
The Polycom® VVX® 500 performance business media phone unifies superior voice capabilities and applications into a simple-to-use, yet high performance unified communications (UC) solution.
The Polycom VVX 500 phone is built for today's busy managers and knowledge workers who need a powerful, expandable office phone that keeps up with their multitasking and schedule-juggling. Building on the behavior common to mobile phones, the multi-touch, gesture-based user interface of the VVX 500 phone makes navigation intuitive and easy.
VVX 500 enhanced productivity, complementing the workplace applications on the user's computer. Users benefit from such capabilities as viewing their Outlook calendar on the phone and receiving meeting reminders while still having access to their corporate directory, all while waiting for their PCs to boot. Users can also extend their PC desktop to include the VVX 500 phone's screen, helping to enable simplified interactions and dialing using their PC's mouse and keyboard.
The VVX 500 phone is ready for future expansion modules and accessories for applications such as video conferencing and even wireless networking. As the most customizable Polycom phone so far, the VVX 500 provides personalized information at a glance, through built-in Web applications and even a digital photo frame. Polycom's "My Info Portal" serves up stocks, news, sports, weather, and other streamed content directly to the phone screen.
The Polycom VVX 500 phone is built for today's busy managers and knowledge workers who need a powerful, expandable office phone that keeps up with their multitasking and schedule-juggling. Building on the behavior common to mobile phones, the multi-touch, gesture-based user interface of the VVX 500 phone makes navigation intuitive and easy.
VVX 500 enhanced productivity, complementing the workplace applications on the user's computer. Users benefit from such capabilities as viewing their Outlook calendar on the phone and receiving meeting reminders while still having access to their corporate directory, all while waiting for their PCs to boot. Users can also extend their PC desktop to include the VVX 500 phone's screen, helping to enable simplified interactions and dialing using their PC's mouse and keyboard.
The VVX 500 phone is ready for future expansion modules and accessories for applications such as video conferencing and even wireless networking. As the most customizable Polycom phone so far, the VVX 500 provides personalized information at a glance, through built-in Web applications and even a digital photo frame. Polycom's "My Info Portal" serves up stocks, news, sports, weather, and other streamed content directly to the phone screen.
Caller ID Show/Restrict
Quick Redial
Volume Keys
Voice/Video Mail
Ringtones
12 Calling Lines
100 or more contacts
3-way Call
Call Forward
Call Hold
Mute Key
3.5" TFT LCD (320 x 240)
Web Browser
Polycom VVX 600
0
Polycom VVX Expansion Module
0
SoundPoint IP Expansion Module
$249.99
SoundStation IP 5000
Enhance productivity and enrich collaboration with the ultimate, all-in-one, one-touch desktop UC solution designed specifically for executives, managers and knowledge workers. The VVX 600 series business media phone delivers a best-in-class personal communications experience with an extensive list of easy-to-use features that complement the way you work. The VVX 600 series is simple for administrators to deploy, maintain and upgrade while seamlessly integrating with third-party productivity applications. Leverage previous IT infrastructure investments and protect your investment into the future with options such as add-on video, wireless and other accessories.
• Enhance executive productivity
• Large (4.3”) TFT (480 x 272) capacitive touch-screen
• Up to 16 line appearances/speed dials
• Hard Keys: 12-key keypad, home, speaker, mute, volume, headset
• Video playback and video conferencing via external USB cam
• Integrated Bluetooth
• Best-in-class personal communications
• Legendary Polycom HD Voice technology up to 14KHz on all audio paths (Speaker, Handset, Headset)
• 2 x Ethernet 10/100/1000
• 2 x USB 2.0 host
• Green – low power
• Enhance executive productivity
• Large (4.3”) TFT (480 x 272) capacitive touch-screen
• Up to 16 line appearances/speed dials
• Hard Keys: 12-key keypad, home, speaker, mute, volume, headset
• Video playback and video conferencing via external USB cam
• Integrated Bluetooth
• Best-in-class personal communications
• Legendary Polycom HD Voice technology up to 14KHz on all audio paths (Speaker, Handset, Headset)
• 2 x Ethernet 10/100/1000
• 2 x USB 2.0 host
• Green – low power
Turn Your Polycom VVX Business Media Phone Into a High-Performance Attendant Console
• 40 illuminated bi-color LED keys programmable line keys
• Daisy-chainable for a total of 120 contacts (3 x 40)
• 2 x AUX ports (SPI) for connectivity and power propagation from the host
• Expandability up to 3 modules
• Legacy PoE support: Up to 3 modules powered by host phone (VVX 300/400/500/600)
• 40 illuminated bi-color LED keys programmable line keys
• Daisy-chainable for a total of 120 contacts (3 x 40)
• 2 x AUX ports (SPI) for connectivity and power propagation from the host
• Expandability up to 3 modules
• Legacy PoE support: Up to 3 modules powered by host phone (VVX 300/400/500/600)
The SoundPoint IP Expansion Module for the SoundPoint IP 601/650 is an optimal solution for telephone attendants "“ receptionists, administrative assistants, secretaries, and other "œpower users" "“ who manage and monitor multiple simultaneous telephone calls on a daily basis. The enhanced user interface and advanced call handling capabilities help boost telephone attendant productivity, while enabling an easy transition from traditional PBX features and functionality to Voice over IP.
With a high-performance attendant console based on the SoundPoint IP 601 and up to three SoundPoint IP Expansion Modules telephone attendants can promptly accept, accurately screen, efficiently dispatch, and effortlessly monitor calls to reduce the number of lost calls, shorten transaction times, and increase the accuracy of call routing.
With a high-performance attendant console based on the SoundPoint IP 601 and up to three SoundPoint IP Expansion Modules telephone attendants can promptly accept, accurately screen, efficiently dispatch, and effortlessly monitor calls to reduce the number of lost calls, shorten transaction times, and increase the accuracy of call routing.
The Polycom SoundStation IP 5000 conference phone delivers remarkably clear conference calls for small conference rooms and executive offices. It features Polycom HD Voice™ technology, broad SIP interoperability, and a modern design that is ideal for smaller rooms—all at an affordable price.
With Polycom HD Voice technology, the SoundStation IP 5000 conference phone boosts productivity and reduces listener fatigue by turning ordinary conference calls into crystal-clear, interactive conversations. It captures both the deeper lows and higher frequencies of the human voice for conference calls that sound as natural as being there.
For all calls, the SoundStation IP 5000 conference phone delivers advanced audio performance that is designed for executive offices and smaller conference rooms with up to 6 participants. From full-duplex technology that eliminates distracting drop-outs to the latest echo cancellation advancements, only Polycom can deliver a conference phone experience with no compromises. Conference calls are made more productive and efficient by three sensitive microphones with 360° coverage that allow users to speak in a normal voice and be heard clearly from up to 7 feet away.
With Polycom HD Voice technology, the SoundStation IP 5000 conference phone boosts productivity and reduces listener fatigue by turning ordinary conference calls into crystal-clear, interactive conversations. It captures both the deeper lows and higher frequencies of the human voice for conference calls that sound as natural as being there.
For all calls, the SoundStation IP 5000 conference phone delivers advanced audio performance that is designed for executive offices and smaller conference rooms with up to 6 participants. From full-duplex technology that eliminates distracting drop-outs to the latest echo cancellation advancements, only Polycom can deliver a conference phone experience with no compromises. Conference calls are made more productive and efficient by three sensitive microphones with 360° coverage that allow users to speak in a normal voice and be heard clearly from up to 7 feet away.
SoundStation IP 6000
$969.99
SoundStation IP 7000
$1219.99
Yealink W52P Cordless
$199.99
The SoundStation IP 6000 is an advanced IP conference phone that delivers superior performance for small to midsize conference rooms. The SoundStation IP 6000 features Polycom HD Voice technology, boosting productivity and reducing listener fatigue by turning ordinary conference calls into crystal-clear interactive conversations. It delivers high-fidelity audio from 220 Hz to 14 kHz, capturing both the deeper lows and higher frequencies of the human voice for conference calls that sound as natural as being there.
For all conference calls, the SoundStation IP 6000 delivers advanced audio performance that far exceeds previous generations of conference phones. From full-duplex technology that eliminates distracting drop-outs to the latest echo cancellation advancements, only Polycom can deliver a conference phone experience with no compromises. Plus, Automatic Gain Control intelligently adjusts the microphone sensitivity based on where participants are seated in the conference room, making the conversations clearer for all participants. It also features technology that resists interference from mobile phones and other wireless devices, delivering clear communications without distractions.
For all conference calls, the SoundStation IP 6000 delivers advanced audio performance that far exceeds previous generations of conference phones. From full-duplex technology that eliminates distracting drop-outs to the latest echo cancellation advancements, only Polycom can deliver a conference phone experience with no compromises. Plus, Automatic Gain Control intelligently adjusts the microphone sensitivity based on where participants are seated in the conference room, making the conversations clearer for all participants. It also features technology that resists interference from mobile phones and other wireless devices, delivering clear communications without distractions.
The SoundStation IP 7000 is a breakthrough conference phone that delivers outstanding performance and a robust feature set for SIP-based VoIP platforms. It is the most advanced conference phone ever developed, and is ideal for executive offices, conference rooms, and board rooms. The SoundStation IP 7000 features Polycom HD Voice technology, boosting productivity and reducing listener fatigue by turning ordinary conference calls into crystal-clear interactive conversations. It delivers high-fidelity audio from 160 Hz to 22 kHz, capturing both the deeper lows and higher frequencies of the human voice for conference calls that sound as natural as being there. For all conference calls, the SoundStation IP 7000 delivers advanced audio performance that far exceeds previous generations of conference phones. From full-duplex technology that eliminates distracting drop-outs to the latest echo cancellation advancements, only Polycom can deliver a conference phone experience with no compromises.
Yealink W52P is a SIP Cordless Phone System designed for small business and SoHo who are looking for immediate cost saving but scalable SIP-based mobile communications system.Combining the benefits of wireless communication with rich business features of Voice over IP telephony, User can benefit from freedom of movement, lifelike voice communications, multi-tasking convenience, professional features like intercom, transfer, call forward, 3-way conferencing, PoE etc. This system works with widely-known Broadsoft, Asterisk, 3CX and supports quick and easy configuration.
2102 Residential VoIP Access Device
$99.99 ($4/mo)
Mediatrix 4102S
0
Mediatrix 4104
$279.99
Mediatrix 4108
$599.99
The Mediatrix 2102 is a high-quality and cost-efficient VoIP gateway.
The Mediatrix 2102 can be deployed as a residential IP telephony access device, enabling service providers to cost-effectively deliver VoIP services to their subscriber base. The Mediatrix 2102 connects up to two analog phones and/or faxes, as well as a PC, to a service provider's network over a single broadband connection.
Through a 10/100 BaseT EthernetWAN interface, the Mediatrix 2102 connects analog terminals and a PC directly to a broadband modem without the need for an external router, and with only a single IP address provided by the service provider. With an embedded PPPoE client and its innovative IP technology, the Mediatrix 2102 and the PC connected to the second Ethernet port have the same public IP address, without private IP addresses or any address translation necessary.
The Mediatrix 2102 can be deployed as a residential IP telephony access device, enabling service providers to cost-effectively deliver VoIP services to their subscriber base. The Mediatrix 2102 connects up to two analog phones and/or faxes, as well as a PC, to a service provider's network over a single broadband connection.
Through a 10/100 BaseT EthernetWAN interface, the Mediatrix 2102 connects analog terminals and a PC directly to a broadband modem without the need for an external router, and with only a single IP address provided by the service provider. With an embedded PPPoE client and its innovative IP technology, the Mediatrix 2102 and the PC connected to the second Ethernet port have the same public IP address, without private IP addresses or any address translation necessary.
The Mediatrix 4102S SIP and DGW 2.0 VoIP gateway is a high-quality and cost efficient VoIP gateway and analog telephone adapter (ATA) which connects homes, larger branch offices or multi-tenant buildings to an IP network, while preserving investment in analog telephones and faxes. It features 2 FXS ports for analog devices such as phones, modems, or fax machines, and a WAN and LAN RJ45 connection.
The Mediatrix 4100 enables cost-effective VoIP deployments in medium-size branch offices and multi-tenant applications. The Mediatrix 4100 has the additional benefit of supporting high compression codec's simultaneously on each analog voice ports, thus saving valuable bandwidth. As all other Mediatrix devices, the 4100 Series provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end users. To further facilitate deployments, factory loaded configurations are possible.
The Mediatrix 4100 offers security features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects. In addition, an intelligent PSTN bypass allows Mediatrix 4100 users to make emergency calls and maintain their phone service in the event of a power outage or network failure.
• 2 FXS RJ11 Ports
• 1 RJ45 LAN, 1 RJ45 WAN
• Carrier-grade voice quality
• Fax over IP support, including T.38
• Automatic firmware and configuration file download
• SNMP and web management
• TFTP or HTTP auto-provisioning
• Support for SNMPv3
• Encrypted configuration files support
• HTTP Digest authentication
• Compliant with multiple enhanced security protocols offering a rich feature set including: SIP, MIKEY, TLS, SRTP, certificates management, and HTTPS
• QoS features support
• DHCP client
• STUN Client
The Mediatrix 4100 enables cost-effective VoIP deployments in medium-size branch offices and multi-tenant applications. The Mediatrix 4100 has the additional benefit of supporting high compression codec's simultaneously on each analog voice ports, thus saving valuable bandwidth. As all other Mediatrix devices, the 4100 Series provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end users. To further facilitate deployments, factory loaded configurations are possible.
The Mediatrix 4100 offers security features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects. In addition, an intelligent PSTN bypass allows Mediatrix 4100 users to make emergency calls and maintain their phone service in the event of a power outage or network failure.
• 2 FXS RJ11 Ports
• 1 RJ45 LAN, 1 RJ45 WAN
• Carrier-grade voice quality
• Fax over IP support, including T.38
• Automatic firmware and configuration file download
• SNMP and web management
• TFTP or HTTP auto-provisioning
• Support for SNMPv3
• Encrypted configuration files support
• HTTP Digest authentication
• Compliant with multiple enhanced security protocols offering a rich feature set including: SIP, MIKEY, TLS, SRTP, certificates management, and HTTPS
• QoS features support
• DHCP client
• STUN Client
The Mediatrix 4100 Series include the Mediatrix 4102, 4104, 4108, 4116 and the 4124. The Mediatrix 4100 Series connects up to 24 analog phones and/or faxes to a broadband modem or LAN.
The Mediatrix 4100 Series access devices are high-quality, cost efficient VoIP gateways connecting small to large branch offices and multi-tenant buildings to an IP network, while preserving existing investments in analog telephones and faxes.
The Mediatrix 4100 Series allow Service Providers to deploy rapidly and economically their solutions in medium-size enterprises as well as being the ideal solution for branch office connectivity to larger private networks.
The Mediatrix 4100 Series has the additional benefit of supporting high compression codec's simultaneously on each analog voice ports, thus saving valuable bandwidth. In addition The Mediatrix 4100 Series offers features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects.
In addition, an intelligent PSTN bypass allows Mediatrix 4100 users to make emergency calls and maintain their phone service in the event of a power outage or network failure.
The Mediatrix 4100 Series also provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible.
The Mediatrix 4100 Series access devices are high-quality, cost efficient VoIP gateways connecting small to large branch offices and multi-tenant buildings to an IP network, while preserving existing investments in analog telephones and faxes.
The Mediatrix 4100 Series allow Service Providers to deploy rapidly and economically their solutions in medium-size enterprises as well as being the ideal solution for branch office connectivity to larger private networks.
The Mediatrix 4100 Series has the additional benefit of supporting high compression codec's simultaneously on each analog voice ports, thus saving valuable bandwidth. In addition The Mediatrix 4100 Series offers features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects.
In addition, an intelligent PSTN bypass allows Mediatrix 4100 users to make emergency calls and maintain their phone service in the event of a power outage or network failure.
The Mediatrix 4100 Series also provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible.
The Mediatrix 4100 Series include the Mediatrix 4102, 4104, 4108, 4116 and the 4124. The Mediatrix 4100 Series connects up to 24 analog phones and/or faxes to a broadband modem or LAN.
The Mediatrix 4100 Series access devices are high-quality, cost efficient VoIP gateways connecting small to large branch offices and multi-tenant buildings to an IP network, while preserving existing investments in analog telephones and faxes.
The Mediatrix 4100 Series allow Service Providers to deploy rapidly and economically their solutions in medium-size enterprises as well as being the ideal solution for branch office connectivity to larger private networks.
The Mediatrix 4100 Series has the additional benefit of supporting high compression codec's simultaneously on each analog voice ports, thus saving valuable bandwidth. In addition The Mediatrix 4100 Series offers features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects.
The Mediatrix 4100 Series also provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible.
The Mediatrix 4100 Series access devices are high-quality, cost efficient VoIP gateways connecting small to large branch offices and multi-tenant buildings to an IP network, while preserving existing investments in analog telephones and faxes.
The Mediatrix 4100 Series allow Service Providers to deploy rapidly and economically their solutions in medium-size enterprises as well as being the ideal solution for branch office connectivity to larger private networks.
The Mediatrix 4100 Series has the additional benefit of supporting high compression codec's simultaneously on each analog voice ports, thus saving valuable bandwidth. In addition The Mediatrix 4100 Series offers features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects.
The Mediatrix 4100 Series also provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible.
Mediatrix 4116
$1089.99
Mediatrix 4124
$1349.99
Mediatrix C710
0
Mediatrix C711
0
The Mediatrix 4100 Series include the Mediatrix 4102, 4104, 4108, 4116 and the 4124. The Mediatrix 4100 Series connects up to 24 analog phones and/or faxes to a broadband modem or LAN.
The Mediatrix 4100 Series access devices are high-quality, cost efficient VoIP gateways connecting small to large branch offices and multi-tenant buildings to an IP network, while preserving existing investments in analog telephones and faxes.
The Mediatrix 4100 Series allow Service Providers to deploy rapidly and economically their solutions in medium-size enterprises as well as being the ideal solution for branch office connectivity to larger private networks.
The Mediatrix 4100 Series has the additional benefit of supporting high compression codec's simultaneously on each analog voice ports, thus saving valuable bandwidth. In addition The Mediatrix 4100 Series offers features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects.
In addition, an intelligent PSTN bypass allows Mediatrix 4100 users to make emergency calls and maintain their phone service in the event of a power outage or network failure.
The Mediatrix 4100 Series also provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible.
The Mediatrix 4100 Series access devices are high-quality, cost efficient VoIP gateways connecting small to large branch offices and multi-tenant buildings to an IP network, while preserving existing investments in analog telephones and faxes.
The Mediatrix 4100 Series allow Service Providers to deploy rapidly and economically their solutions in medium-size enterprises as well as being the ideal solution for branch office connectivity to larger private networks.
The Mediatrix 4100 Series has the additional benefit of supporting high compression codec's simultaneously on each analog voice ports, thus saving valuable bandwidth. In addition The Mediatrix 4100 Series offers features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects.
In addition, an intelligent PSTN bypass allows Mediatrix 4100 users to make emergency calls and maintain their phone service in the event of a power outage or network failure.
The Mediatrix 4100 Series also provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible.
The Mediatrix 4100 Series include the Mediatrix 4102, 4104, 4108, 4116 and the 4124. The Mediatrix 4100 Series connects up to 24 analog phones and/or faxes to a broadband modem or LAN.
The Mediatrix 4100 Series access devices are high-quality, cost efficient VoIP gateways connecting small to large branch offices and multi-tenant buildings to an IP network, while preserving existing investments in analog telephones and faxes.
The Mediatrix 4100 Series allow Service Providers to deploy rapidly and economically their solutions in medium-size enterprises as well as being the ideal solution for branch office connectivity to larger private networks.
The Mediatrix 4100 Series has the additional benefit of supporting high compression codec's simultaneously on each analog voice ports, thus saving valuable bandwidth. In addition The Mediatrix 4100 Series offers features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects.
In addition, an intelligent PSTN bypass allows Mediatrix 4100 users to make emergency calls and maintain their phone service in the event of a power outage or network failure.
The Mediatrix 4100 Series also provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible.
The Mediatrix 4100 Series access devices are high-quality, cost efficient VoIP gateways connecting small to large branch offices and multi-tenant buildings to an IP network, while preserving existing investments in analog telephones and faxes.
The Mediatrix 4100 Series allow Service Providers to deploy rapidly and economically their solutions in medium-size enterprises as well as being the ideal solution for branch office connectivity to larger private networks.
The Mediatrix 4100 Series has the additional benefit of supporting high compression codec's simultaneously on each analog voice ports, thus saving valuable bandwidth. In addition The Mediatrix 4100 Series offers features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects.
In addition, an intelligent PSTN bypass allows Mediatrix 4100 users to make emergency calls and maintain their phone service in the event of a power outage or network failure.
The Mediatrix 4100 Series also provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible.
The Mediatrix C710 VoIP Gateway is a business-class analog gateway featuring 4 FXS ports and 2 10/100 Ethernet ports. The Mediatrix C710's 4 FXS ports allow you to use analog phones or fax machines on a VoIP system, or allow you to connect a legacy analog PBX to a VoIP Service Provider so you can save money with VoIP without replacing your legacy PBX. The Mediatrix C710 supports industry standard codecs such as G.711a/u, G.723, G.726, and G.729a/u.
The Mediatrix C7 enables cost-effective VoIP deployments in small offices for both IP Centrex and private network applications. It targets enterprises with brand offices or remote workers, SMB's seeking to realize the cost savings of VoIP and SIP trunking without upgrading their legacy phone equipment, and businesses looking to migrate to VoIP slowly with a migration transparent to its users.
The Mediatrix C7 series VoIP gateway offers more than just connectivity. The C710 supports QoS features and includes an integrated VoIP-enabled IP router.
• 4 FXS Ports
• 2 x 10/100 Ethernet Ports
• North American Power Supply included
• DSP supporting a minimum of 8 bi-directional channels of encrypted and compressed media
• SIP – RFC3261
• Dual-Stack IPv4/IPv6 support using ANAT (RFC 4091)
• Vocoders: G.711 (A-law, μ-law), G.723.1, G.726, G.729a/b
• G.168 echo cancellation (64 ms)
• HTTPS, for web pages and for exchange of Configuration File
• SRTP with MIKEY and SDES: Supported Cypher, AES – 128 bits
• Call Forward / Call Transfer / Conference Call / Call Waiting support
• Echo Cancellation / Dynamic Jitter Buffer / Voice Activity Detection / Silence Suppression
• Message Waiting Indication, via FSK
• Flash hook event signaling
• Caller ID Generation (Name & Number) as per Bellcore DTMF or FSK
The Mediatrix C7 enables cost-effective VoIP deployments in small offices for both IP Centrex and private network applications. It targets enterprises with brand offices or remote workers, SMB's seeking to realize the cost savings of VoIP and SIP trunking without upgrading their legacy phone equipment, and businesses looking to migrate to VoIP slowly with a migration transparent to its users.
The Mediatrix C7 series VoIP gateway offers more than just connectivity. The C710 supports QoS features and includes an integrated VoIP-enabled IP router.
• 4 FXS Ports
• 2 x 10/100 Ethernet Ports
• North American Power Supply included
• DSP supporting a minimum of 8 bi-directional channels of encrypted and compressed media
• SIP – RFC3261
• Dual-Stack IPv4/IPv6 support using ANAT (RFC 4091)
• Vocoders: G.711 (A-law, μ-law), G.723.1, G.726, G.729a/b
• G.168 echo cancellation (64 ms)
• HTTPS, for web pages and for exchange of Configuration File
• SRTP with MIKEY and SDES: Supported Cypher, AES – 128 bits
• Call Forward / Call Transfer / Conference Call / Call Waiting support
• Echo Cancellation / Dynamic Jitter Buffer / Voice Activity Detection / Silence Suppression
• Message Waiting Indication, via FSK
• Flash hook event signaling
• Caller ID Generation (Name & Number) as per Bellcore DTMF or FSK
The Mediatrix C711 VoIP Gateway is a business-class analog gateway featuring 8 FXS ports and 2 10/100 Ethernet ports. The Mediatrix C711's 8 FXS ports allow you to use analog phones or fax machines on a VoIP system, or allow you to connect a legacy analog PBX to a VoIP Service Provider so you can save money with VoIP without replacing your legacy PBX. The Mediatrix C711 supports industry standard codecs such as G.711a/u, G.723, G.726, and G.729a/u.
The Mediatrix C7 enables cost-effective VoIP deployments in small offices for both IP Centrex and private network applications. It targets enterprises with brand offices or remote workers, SMB's seeking to realize the cost savings of VoIP and SIP trunking without upgrading their legacy phone equipment, and businesses looking to migrate to VoIP slowly with a migration transparent to its users.
The Mediatrix C7 series VoIP gateway offers more than just connectivity. The C711 supports QoS features and includes an integrated VoIP-enabled IP router.
• 8 FXS Ports
• 2 x 10/100 Ethernet Ports
• North American Power Supply included
• DSP supporting a minimum of 8 bi-directional channels of encrypted and compressed media
• SIP – RFC3261
• Dual-Stack IPv4/IPv6 support using ANAT (RFC 4091)
• Vocoders: G.711 (A-law, μ-law), G.723.1, G.726, G.729a/b
• G.168 echo cancellation (64 ms)
• HTTPS, for web pages and for exchange of Configuration File
• SRTP with MIKEY and SDES: Supported Cypher, AES – 128 bits
• Call Forward / Call Transfer / Conference Call / Call Waiting support
• Echo Cancellation / Dynamic Jitter Buffer / Voice Activity Detection / Silence Suppression
• Message Waiting Indication, via FSK
• Flash hook event signaling
• Caller ID Generation (Name & Number) as per Bellcore DTMF or FSK
The Mediatrix C7 enables cost-effective VoIP deployments in small offices for both IP Centrex and private network applications. It targets enterprises with brand offices or remote workers, SMB's seeking to realize the cost savings of VoIP and SIP trunking without upgrading their legacy phone equipment, and businesses looking to migrate to VoIP slowly with a migration transparent to its users.
The Mediatrix C7 series VoIP gateway offers more than just connectivity. The C711 supports QoS features and includes an integrated VoIP-enabled IP router.
• 8 FXS Ports
• 2 x 10/100 Ethernet Ports
• North American Power Supply included
• DSP supporting a minimum of 8 bi-directional channels of encrypted and compressed media
• SIP – RFC3261
• Dual-Stack IPv4/IPv6 support using ANAT (RFC 4091)
• Vocoders: G.711 (A-law, μ-law), G.723.1, G.726, G.729a/b
• G.168 echo cancellation (64 ms)
• HTTPS, for web pages and for exchange of Configuration File
• SRTP with MIKEY and SDES: Supported Cypher, AES – 128 bits
• Call Forward / Call Transfer / Conference Call / Call Waiting support
• Echo Cancellation / Dynamic Jitter Buffer / Voice Activity Detection / Silence Suppression
• Message Waiting Indication, via FSK
• Flash hook event signaling
• Caller ID Generation (Name & Number) as per Bellcore DTMF or FSK