The handset features peer-to-peer dialing, speed dial, 3-way conferencing, call waiting, call transfer, and call forward, mute, hold and selectable ringtones. The large, full-color high resolution LCD display features an intuitive user interface enabling users to easily and quickly configure the handset when traveling within range of other Wireless-G networks so you can make VoIP calls wherever you go.

You can also surf the Internet with the built-in web browser and can even receive live video from anywhere in the world and view it right on your WIP330 Wireless-G IP Phone when you access any web camera, like the WVC54GC Wireless-G Internet Video Camera, also from Linksys!

Handset stores the last 20 call history records and can save 250 phone book entries. Personalize your phone with a selection of ringtones that reflects your style.

• Support SIP v2 Standards
• Compliance with IEEE 802.11b/g Wireless Standard
• Powered by Microsoft Windows CE, with IE Web browser
• High Resolution Color LCD screen
• Support QoS (Quality of Service) to ensure best quality voice
• Enhanced Power Saving Design for Extended Standby and Talk Time
• 50 hours standby time, 3 hours talk time on average
• 3-Way Conferecing, Call Hold and Resume, and Caller ID
• Fast Hotspot Authentication
• Supports auto-provisioning using HTTP or HTTPS for configuration and upgrades
• Outdoor up to 300m via Embedded Antenna
• One mini-USB Socket, One Stereo Ear Phone Jack
• QVGA TIF 2.2 inch LCD (240*320 pixels) with 65K colors
• 32MB Flash, 64MB SDRAM
• Multiple Access point Registration Support
• TCP/UDP/IP, IPV4, DNS, SDP, ARP, ICMP, DHCP Client, Static IP
• WEP (64/128), WPA-PSK Encryption
• ToS
• SIP v2 Session Initiation Protocol (RFC3261), SDP (RFC2327)
• SIP Session Keep Alive
• G.711( A-law and μ-law), G.729 A
• In-band, Out-band (RFC2833)
• G.168 Echo Cancellation
• Jitter Buffer Control – (default 180ms, max 900ms)
• Comfort Noise Generation
• Packet Loss Concealment
• Speaker and Microphone Volume Control
• VAD – Voice Activity Detection
• 3-Way Conferencing
• Peer-to-Peer Dialing
• Call Hold and Resume
• Caller ID Presentation
• Caller ID Presentation Restriction
• Dial by Phone Number
• Call Forward
• DTMF Tone Detection
• Consultation Hold and Transfer
• Call Waiting and Retrieve
• Mute
• Speed Dial
• Last Number Redial
• Volume Control
• Ringtones: True Tones
• Phone Book (250 records)
• Call History (20 Records)
• Language (English/Spainish)
• Vibrator (Silent mode)
• Password Security
• Date & Time (NTP time synchronization)
• Internet Web Browser (Microsoft IE)
• AES or SSL Encryption
• Firmware upgrades using HTTP, or HTTPS
• Configuration change using HTTP, or HTTPS
• Embedded Web configuration interface (with password protection)
• Power-up Diagnostic

Features

• Easy configuration of internet telephony (VoIP) with or without a PC
• Up to 3 calls in parallel: 2 VoIP calls and 1 fixed-line call with multiple handsets
• Large illuminated display with 65,000 colors and 128 x 160 pixels
• Easy access to online information via screensaver and interactive Info Center
• Includes an integrated answering machine
• Increased virus protection thanks to protected operating system
• Convenient handsfree talking
• SMS with up to 640 characters
• Up to 250 vCard entries to store first name, last name, 3 numbers, e-mail address and birthday of your contacts

Features

• Six lines (standalone)/34 lines with three Color Expansion Modules
• Backlit 320 x 160 pixel color display delivers a rich visual experience
• Integrated dual-port Gigabit Ethernet switch
• XHTML microbrowser for Polycom and third-party applications
• Advanced SIP functionality, including shared lines, and busy lamp field
• Broad and robust interoperability with a broad array of SIP call control platforms
• Hands-free speakerphone
• Built-in USB port to support applications

Features

• Up to 4 lines with up to 2 concurrent calls per line
• 320 x 160-pixel backlit grayscale graphical LCD
• Built-in GigE support
• Cutting-edge SIP features
• Supports seamless migration to GigE-based network infrastructure, today or tomorrow
• Advanced functionality, including shared lines, busy lamp field, and XHTML Microbrowser
• Hands-free speakerphone

Features

• Up to 4 lines with up to 2 concurrent calls per line
• Backlit 320 x 160-pixel grayscale graphical LCD
• Support of G.722 wideband codec, Acoustic Clarity Technology 2, and systems design optimized for Polycom HD Voice technology
• Integrated IEEE 802.3af Power over Ethernet (PoE) support
• Advanced functionality, including shared lines, and busy lamp field, and XHTML Microbrowser
• Cutting-edge SIP features
• Hands-free speakerphone
• Interoperability with leading SIP-based IP PBX and Softswitch platforms
• Two-port 10/100 Mbps Ethernet switch

Features

• Up to 3 lines with up to 2 concurrent calls per line
• Backlit 256 x 116-pixel multi-layer, grayscale graphical LCD
• Polycom HD Voice technology, including support of G.722 wideband codec, and systems design optimized for Polycom HD Voice
• Advanced functionality, including shared lines, and busy lamp field, and XHTML Microbrowser
• Integrated IEEE 802.3af Power over Ethernet (PoE) support (Class 2)
• Two port 10/100 Ethernet Switch
• Dedicated RJ-9 headset port
• Hands-free speakerphone

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.

Features

• Six-line, feature-rich phone with Polycom HD Voice
• Intuitive, color touch-screen interface for voice, video and applications
• Instant, one-touch business-grade video conferencing right from the desktop
• Adjustable camera, base, and display to suit the environment and provide eye-level visual interactions
• 7 in TFT LCD display with adjustable screen angle
• On-screen virtual keyboard
• Screen saver and digital picture frame mode
• Integrated Gigabit Ethernet (GigE) switch for bandwidth-intensive applications that supports PoE
• Flexible and Future-Proof Dual stack support ensures easy connectivity to both H.323 video and SIP telephony simultaneously
• Highly customizable applications platform with open Polycom XML API , integrated microbrowser and USB 2.0 for applications
• IEEE 802.3af powered at under 11W maximum for a “green” world
• Smart motion detection to enable the screen to go to power-save mode when no one is in the office
• Multiple call/flexible line appearances
• Called, calling, connected party identification/information
• Deliver a business-grade experience with one personalized device
• Achieve faster decision making with the ability to integrate communications into business processes

International Telecoms Week (ITW) is the meeting point for the international telecoms wholesale community – each year over 4,000 delegates from nearly 1,300 companies and from over 130 countries meet in Washington DC to buy and sell both voice and data products and services. ITW is an event for the entire industry with participants from all areas of the wholesale industry: Tier 1, Tier 2 and Tier 3 carriers, mobile / wireless operators, ISPs, VoIP providers and technology partners from the voice, data, satellite, sub-sea and fixed line markets.

ITW 2010 takes place from Monday 24 May – Wednesday 26 May 2010.

Book a meeting with Iristel Inc.

Headquartered in Markham, Ontario, Canada, IRISTEL Inc. is a leading global provider of wireless voice and data IP services. The company was founded in 1999 to deliver the highest quality Voice over Internet Protocol (VoIP) network and services. IRISTEL is a pioneer in the Canadian VoIP market.

IRISTEL was granted a carrier license by the Canadian Radio-Television and Telecommunications Commission (CRTC) in 2000, and today it is one of the few VoIP operator with the Competitive Local Exchange Carrier (CLEC) designation in Canada. IRISTEL also has an additional 20 points of presence (POPs) around the globe.

As one of the leaders in telecommunications business, IRISTEL is one of the official sponsors of ITW 2010.

If you are interested in booking a meeting at ITW 2010 with the Iristel Inc staff, please use the form below to send us a message.

Iristel has decreased the rate on calls to Toll Free numbers from $0.05/minute to $0.03/minute. This modification will come into effect starting from 1st of January 2010.