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VoIP phones

VoIP ATA

VoIP PBX/Gateway

ISDN Interfases

VoIP PBX (АТС)

Smart Node
 
 
 
 
 
 

VoIP Phone Snom 300

  • Two-line display (2 x 16 characters)
  • 27 keys, 7 LEDs
  • 6 programmable function keys
  • 2 x IEEE 802.3 10/100 Mbps switch
  • 4 identities
  • Headset connection
  • SIP RFC3261
  • Security: SIPS/SRTP, TLS
  • STUN, ENUM, NAT
  • Codecs: G.711, G.729A, G.723.1, G.722, G.726, GSM 6.10 (full rate)
VoIP Phone Snom 320
  • Two-line tiltable semigraphical display
  • 47 keys, 13 LEDs
  • 12 programmable function keys
  • Speakerphone
  • 2 x IEEE 802.3 10/100 Mbps switch
  • Power over Ethernet
  • Headset connection
  • SIP RFC3261
  • Security: SIPS/SRTP, TLS STUN, ENUM, NAT, ICE
  • Codecs: G.711, G.729A, G.723.1, G.722, G.726, GSM 6.10 (full rate)
  • National Language Support
VoIP Phone Snom 360
  • Tiltable graphical display (128 x 64 pixels)
  • 47 keys, 13 LEDs
  • 12 programmable function keys
  • Speakerphone
  • 2 x IEEE 802.3 10/100 Mbps switch
  • Power over Ethernet
  • Headset connection
  • Expansion module with 42 programmable function keys
  • SIP RFC3261
  • Security: SIPS/SRTP, TLS
  • STUN, ENUM, NAT, ICE
  • Codecs: G.711, G.729A, G.723.1, G.722, G.726, GSM 6.10 (full rate)
  • National Language Support
VoIP Phone Snom 370
  • Tiltable high-definition graphical display (240 x 128 pixels)
  • Call indication LED
  • 47 keys, 13 LEDs
  • 12 programmable function keys
  • Speakerphone
  • 2 x IEEE 802.3 10/100 Mbps switch
  • Power over Ethernet
  • Multiple audio device support
  • Expansion module with 42 programmable function keys
  • SIP RFC3261
  • Security: SIPS/SRTP, TLS
  • STUN, ENUM, NAT, ICE
  • Codecs: G.711, G.729A, G.723.1, G.722, G.726, GSM 6.10 (full rate)
  • National Language Support
VoIP Phone Snom M3
  • Display: 128 x 128 pixels, 65536 colors, backlit
  • Li-Ion battery pack for 8 hours of calls or 100 hours standby
  • Range: 50 meters indoors, 100 meters outdoors
  • 12 numerical keys, 5 navigation keys, 2 function keys
  • Speakerphone on mobile handset
  • Polyphonic ringtones
  • Automatic registration of handset
  • Separate charging cradle for handset
  • 8 handsets per base station
  • 8 SIP registrations with different servers/registrars
  • Up to 3 concurrent calls per base station
  • Three-way conference
  • Remote setup, password protection
  • Open DECT GAP standard
Snom Headsets HS-MM2 and HS-MM3
 

The snom HS-MM3 headset is made specifically for use with the snom 300 phone whereas the HS-MM2 headset is used for the snom 320, snom 360, and snom 370 phones.

Headsets allow their users a much wider range of movement than possible with a telephone receiver. This investment is worthwhile for anyone who spends a an office, in a home office, or in a call center - a headset offers many advantages.

 
 
VoIP Phone Snom Expansion Module
 

With the snom 320, 360 and 370 Expansion Module, your snom has the best features to meet the requirements of executive and administrative staff who require high call coverage and flexibility from communication systems.
 
 
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Siemens Gigaset C470IP

Highlights

  • Dual mode: easy switch from internet calls to fixed-line calls by single keypress
  • Information services as screensaver* e.g. weather forecast
  • Expandable phone system with multiline calling for up to 6 handsets and 6 SIP accounts from different providers
  • ECO DECT – for 60% less energy use¹ and variable reduction of transmitting power

Features

  • Easy configuration of internet telephony (VoIP) without a PC
  • Up to 3 calls in parallel: 2 VoIP calls and 1 fixed-line call with multiple handsets
  • E-mail notification with time, date and subject
  • Instant Messaging (buddy list, chatting, presence status)
  • Increased virus protection thanks to protected operating system
  • Convenient handsfree talking in brilliant sound quality
  • Illuminated display with 4096 colors and 128 x 128 pixels
  • Intuitive icon-based menu
  • SMS with up to 612 characters
  • Phonebook for up to 150 names and numbers
 
Siemens optiPoint 150 S SIP (PoE) Siemens optiPoint 150 S SIP (PSU)
The optiPoint 150 S is a cost-effective entry model for Voice over IP telephony. You can access all key features via function keys. It is designed for seamless integration into the Siemens HiPath 8000 communication system and other SIP-based standard platforms for communication solutions in the enterprise and carrier market.

This SIP phone features key functions such as three-party conference, speaker phone, mute, redial, etc. It is simple to operate and easy to administrate. The optiPoint 150 S is ideally suited for environments in which basic business features are required, such as reception areas, corridors or production facilities. The optiPoint 150 S is easy to install and inexpensive to start up. It supports central plug & play deployment and DHCP as well as QoS mechanisms and the STUN protocol.

The optiPoint 150 S features a two-line display and three dialog keys for controlling the various functions (e.g. call forwarding, hold, consultation or call waiting). The two line display also ensures that you can view the name and phone number of the caller. This is one of the reasons why this phone is very popular for business communication in offices, sales rooms and hotels.
 
 
Siemens Enterprise optiPoint WL2

Siemens Enterprise optiPoint WL2 professional S WLAN-Telefon The optiPoint WL2 professional has a backlit color display and offers easy access to a wide range of features. Additional benefits such as an internal phonebook, integration of polyphonic ringtones, voice dialing and access to your company phone directory (LDAP) make the optiPoint WL2 professional the ideal wireless solution.

As one of the best VoWLAN company solutions in its class, the optiPoint WL2 professional offers an unrivaled level of voice quality, a range of security features that surpasses the industry standard, and an easy-to-operate user interface. It also supports the IEEE standards for wireless networking (802.11b) and 54 Mbit/s data transmission (802.11g).

 
 
Siemens Gigaset S675 IP

Highlights

  • Answering machine with up to 30 minutes of recording time
  • High Definition Sound Performance
  • Expandable phone system with multiline functionality for up to 6 handsets and 6 SIP accounts from different providers

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
  • Screensaver to display internet information services
  • Increased virus protection thanks to protected operating system
  • Convenient handsfree talking
  • Large illuminated display with 65,000 colors and 128 x 160 pixels
  • Intuitive icon-based menu
  • 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

 
 
 
 
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VoIP Phone Grandstream GXP-2000

  • Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP
  • Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more
  • Multiline support of up to 11 lines indicators (expandable to a few dozen more through expansion key-module)
  • Graphical LCD to display up to 8 lines and 22 characters per line
  • Dual 10/100Mbps Ethernet ports
  • Headset jack
  • Support Caller ID display or block, per call or permanent
  • Call waiting, Hold, Mute, Transfer (blind or attended), Forward, and more
  • Multi-party conferencing
  • Integrated Power-over-Ether (802.3af)
Grandstream GXP-2000 is a next generation enterprise IP telephone based on open industry standards. Built on innovative technologies, GXP-2000 features market leading superb audio quality, rich functionalities, and excellent manageability at affordable prices.
 
VoIP Phone Grandstream GXP1200
The GXP1200 is a very affordable 2-line entry-level SIP phone that delivers excellent sound quality and advanced feature set. The GXP1200 supports traditional telephony features such as caller id, call waiting, mute, park, hold, transfer and voicemail as well as advanced enterprise features such as 2 independent SIP accounts, multi-language support, 3-way conferencing, and custom ring-tones for distinctive call handling.

The GXP1200 also supports a broad range of voice codecs and features dual 10/100mbps auto-sensing Ethernet ports with integrated Power over Ethernet, headset jack, and secure central configuration file for mass deployment. GUI web interfaces make the GXP1200 quick and easy to install and manage. Broad interoperability with major SIP based platforms ensures compatibility with any IP communications system. Its easy installation, hands-free duplex speakerphone, backlit graphic LCD display, eight dedicated function keys, and XML applications make the GXP1200 a very cost-effective choice for any business that needs a feature rich two line IP phone.
 
 
VoIP Phone Grandstream GXP280

The GXP280 is a next generation small business SIP phone that features 1 line appearance, 128x32 graphic LCD, 3 XML programmable softkeys, and dual 10M/100Mbps network ports. It is an affordable choice for a SOHO user needing a feature rich, 1-line IP phone.

The GXP280 delivers superior audio quality, comprehensive telephony features, automated provisioning, security protection for privacy and broad interoperability with most 3rd party SIP devices, gateways and leading soft-switch platforms. The product has 1 line appearance with FLASH to handle 2 simultaneous calls, offers support for multiple languages (including English, Spanish, German, French, etc) and headset jacks (both 2.5mm and RJ9 jack). Like the other models in the GXP IP phone series, the product is equipped with high fidelity wideband audio and a full-duplex speakerphone with advanced acoustic echo cancellation. The GXP280 is outfitted with SRTP, is wall mountable, and can also be automatically provisioned for mass deployment.
 
 
VoIP Phone Grandstream GXP2010

Technical Features & Benefits

  • Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP
  • Dual switched 10/100Mbps Ethernet ports w/integrated Power-over-Ethernet (802.3af)
  • Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more
  • Backlit 240x120 high resolution graphic LCD with multi-level gray scales

Call Features & Benefits

  • 4 lines indicators with individual SIP account profiles
  • 18 programmable hard keys and 3 XML programmable context-sensitive soft keys
  • Multi-line support of up to 22 call appearance lines with dual-color LED indicators
  • Support Caller ID display or block, per call or permanent
  • Call waiting, hold, mute, transfer (blind or attended), forward, and more
  • Multi-party conferencing (up to 4-way) and shared line appearances
  • Expandable through expansion key-modules (pending)
  • And many more enterprise grade features

Handset Features & Benefits

  • Full duplex speakerphone w/advanced audio echo cancellation
  • Headset jack (2.5 mm and RJ11) for broad headset compatibility
  • Secure and automated provisioning for mass deployment
  • SRTP and TLS (pending) for privacy protection
  • And many more enterprise grade features
 
VoIP Phone Grandstream GXP2020

Technical Features & Benefits

  • Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP
  • Dual switched 10/100Mbps Ethernet ports w/integrated Power-over-Ethernet (802.3af)
  • Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more
  • Backlit 320x160 high resolution graphic LCD with multi-level grey scales

Call Features & Benefits

  • 6 lines indicators with individual SIP account profiles
  • 7 programmable hard keys and 4 XML programmable context-sensitive soft keys
  • Multi-line support of up to 13 call appearance lines with dual-color LED indicators
  • Expandable to additional 112 lines through expansion key-modules
  • Support Caller ID display or block, per call or permanent
  • Call waiting, hold, mute, transfer (blind or attended), forward, and more
  • Multi-party conferencing (up to 5-way)
  • And many more enterprise grade features

Handset Features & Benefits

  • Full duplex speakerphone w/advanced audio echo cancellation
  • Headset jack (2.5 mm and RJ11) for broad headset compatibility
  • Secure and automated provisioning for mass deployment
  • SRTP and TLS (pending) for privacy protection
  • And many more enterprise grade features
VoIP Phone Grandstream BT 200
  • 2x RJ45 10/100Base-T
  • Support SIP 2.0, TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, PPPoE protocols
  • Support NAT traversal via STUN & symmetric RTP
  • Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
  • Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
  • Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
  • Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726,G.728, wide-band G.722 and iLBC. Dynamic negotiation of codec and voice payload length
  • Support standard voice features such as Caller ID Display or Block, Call Waiting, Hold, Transfer, Forward, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, early dial, click-to-dial
  • ression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
  • Support DIGEST authentication and encryption using MD5 and MD5-sess
  • Provide easy configuration thru manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
  • Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
  • NAT-friendly remote software upgrade capability (via tftp and http) even from behind firewalls/NATs.
  • Downloadable music ringing tones
  • FCC - CE - C-tick - RoHS

The Grandstream Budgetone 2000 IP Phone is an award-winning next generation IP network telephone based on industry open standards. Built upon innovative technology, Grandstream IP Phone features market leading superb sound quality and rich functionalities at ultra-affordable price.

Grandstream Expansion Module
  • 56 buttons (each with dual-color LED) that are fully programmable
  • GXP-2000EXT is connected to and controlled by GXP-2000 via a short cable that comes with the package
  • Up to 2 GXP-2000EXT modules can be cascaded with GXP2000
  • essentially allowing up to 112 fully programmable phone extensions
  • support speed dialing, BLF (Busy Lamp Field), call transfer/forward/pickup on each of the programmable buttons on GXP-2000EXT module.
 
 
 
 
 
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Polycom SoundPoint IP 320 Polycom SoundPoint IP 330


Excellent Sound Quality And An Enterprise-Grade Feature Set
  • Two-line entry-level phones
  • Superb sound quality and full-duplex speakerphone performance with Polycom Acoustic Clarity Technology
  • Enterprise-grade feature set
  • Easy to configure and use
  • Integrated Power over Ethernet support
  • Interoperability with leading IP PBX and Softswitch platforms
  • SIP
Features and Benefits
  • Two lines
  • Full-duplex IEEE 1329 Type 1-compliant speakerphone with Polycom Acoustic Clarity Technology
  • 102 x 33-pixel graphical LCD
  • Integrated 802.3af PoE support
  • Two-port 10/100 Ethernet switch - SoundPoint IP 330 (Single 10/100 Ethernet port - SoundPoint IP 320)
  • Small footprint
 
 
 
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LINKSYS SPA921

1-Line IP Telephone with 1 Ethernet Port and Hi-Res Display
Advanced, Affordable, Feature Rich IP Phone for the Home Office and Business
SPA921

  • Full featured one-line business class IP Phone
  • Connect directly to an Internet Telephone Service Provider or connect to an IP PBX
  • Speakerphone. Caller ID. Call Hold, Transfer, Conferencing, and more
  • Easy installation with secure remote provisioning. Menu based and web based configuration.
 
 
LINKSYS SPA922

1-Line IP Telephone with 2-Port Ethernet Switch, PoE and Hi-Res Display
Advanced, Affordable, Feature Rich IP Phone for the Home Office and Business
SPA922

  • Full featured one-line business class IP Phone supporting Power over Ethernet 802.3af
  • Connect directly to an Internet Telephone Service Provider or connect to an IP PBX
  • Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, Conferencing, and more
  • Easy installation and secure remote provisioning. Menu based and web based configuration.
  • An optional power supply is orderable separately for this product if not utilizing Power over Ethernet (PoE)
 
 
LINKSYS SPA941
 4-Line IP Telephone with 1 Ethernet Port and Hi-Res Display
Advanced, Affordable, Feature Rich IP Phone for the Home Office and Business
SPA941
  • Affordable and full featured two or four line business class IP Phone
  • Connect directly to an Internet Telephone Service Provider or connect to an IP PBX
  • Up to four lines. Speakerphone, Caller ID, Call Hold, Transfer, Conferencing, and more
  • Easy installation and secure remote provisioning. Menu based and web based configuration.
 
LINKSYS SPA942

4-Line IP Telephone with 2 Port Ethernet Switch, PoE and Hi-Res Display
Advanced, Affordable, Feature Rich IP Phone for the Home Office and Business
SPA942

  • Full featured four-line business class IP Phone supporting Power over Ethernet 802.3af
  • Connect directly to an Internet Telephone Service Provider or connect to an IP PBX
  • Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, Conferencing, and more
  • Easy installation and secure remote provisioning. Menu based and web based configuration.
  • An optional power supply is orderable separately for this product if not utilizing Power over Ethernet(PoE)
 
LINKSYS SPA962
 6-Line IP Telephone with 2 Port Ethernet Switch, PoE, and Color Display
Enhances Business Communication Requirements with Color
SPA962
  • Full featured six line business class IP Phone supporting Power over Ethernet 802.3af
  • Connect directly to an Internet Telephone Service Provider or connect to an IP PBX
  • Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, Conferencing, and more
  • Appealing Four Inch, True Color Liquid Crystal Display (LCD)
 
LINKSYS WIP-330 Wireless-G
 Stylish, Full-Featured Mobile WiFi Phone for VoIP Service
WIP330
  • Make Low-Cost High Quality VoIP Calls From Wireless Hotspots
  • Easy Set-Up on WiFi Network
  • Superior Range and Performance with 802.11g
  • Intuitive User Interface
 
 
 
 
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VoIP Phone Thomson SpeedTouch 2022

  • 144x32 full graphic display for 3 lines + icons
  • Connectivity: Integrated 2 port 10/100 Ethernet switch with
  • PoE and VLAN tagging
  • Phone services: 2 lines, Call Forward, Call transfer, Call Hold, Call Waiting
  • Handsfree
  • Alert (missed calls, Message Waiting Indicator)
  • Phone book: Local (100 entries)
  • Caller ID Display
  • Automatic Callback
  • 3-ways Conferencing
  • Downloadable rings
  • Multiple power options: power over Ethernet 802.3af and
  • external power supply adapted (included)
  • VoIP Standard: SIP or MGCP
  • Voice compression standards: G.711, G.723, G.729ab
  • IP addressing: Static or dynamic IP configuration (integrated (DHCP client)
  • Quality of Service: ToS DiffServ, 802.1p/Q, VAD, CNG, Packet Loss Compensation, Adaptative Jitter Buffer
  • Web browser interface for configuration and firmware (admin & user mode)
  • Automatic provisioning system (DHCP, TFTP/HTTP)
  • 7 languages
   
VoIP Phone Thomson SpeedTouch 2030
  • 128*64 full graphic display
  • Connectivity: Integrated two port 10/100 Ethernet switch
  • Multilines, Call Forward, Call Transfer, Call Hold, Call Waiting, Redial
  • Handsfree
  • Alert (missed calls, Message Waiting Indicator)
  • 10 Speedial keys/ Supervision backlighted keys (MGCP)
  • Phonebook (100 entries)
  • Caller ID display
  • Automatic Callback
  • Audio extension connection: Integrated headset RJ9 port
    3 ways conferencing
  • Extension module (28 additional backlighted keys)
  • Polyphonic rings
  • Multiple power options: power over Ethernet 802.3af and external power supply adapter
  • VoIP Standard: SIP V2 (RFC 3261) or MGCP
  • Voice compression standards: G.711, G.723, G.729ab
  • IP addressing: Static or Dynamic IP configuration (integrated DHCP client)
  • Quality of Service: ToS Diffserv, 802.1p/Q, VAD, CNG,
    Packet Loss Compensation, Adaptative Jitter Buffer
  • Web browser interface for configuration and firmware
    upgrade (admin & user mode)
  • SNMP
  • Automatic Provisioning System (DHCP, TFTP/HTTP)
  • 5 Languages
 
VoIP Phone Thomson SpeedTouch 2020
  • 2 lines LCD display
  • Integrated 2 port 10/100 Ethernet switch
  • Call Forward, Call Transfer (blind), Call Hold, Redial
    Group listening
  • Message Waiting Indicator
  • Speedial
  • Phonebook
  • Caller ID display
  • Audio extension connection: Integrated headset RJ9 port
  • Multiple power options: power over Ethernet 802.3af and
  • xternal power supply adapter
  • VoIP Standard: SIP V2 (RFC 3261)
  • Voice compression standards: G.711, G.723, G.729AB
  • IP addressing: Static or dynamic IP configuration (integrated DHCP client)
  • Quality of Service: ToS DiffServ, 802.1p/Q, VAD, CNG,
    Packet Loss Compensation
  • Web browser interface for configuration and firmware
    upgrade (admin & user mode)
   
VoIP Phone Thomson SpeedTouch 2030 Extension
  • 1 Module: 28 programmable keys
  • 2 Modules: 56 programmable keys
  • One RJ45 Ethernet to connect ST2030
  • One RJ45 Ethernet to connect to second extension module
  • Short RJ45 Line cord
  • Powered from ST 2030 IP phone via a delivered RJ45 cable
  • Weight: 200 g
  • Size : 9.8 cm x 19.5 cm x 3 cm

The ST2030 extension module is the ideal tool for supporting operators or assistants to monitor additional lines or pre-program function keys. Access to 28 additional programmable keys to make more comfortable the use of your IP phone, you can also
attached a second extra module for more additional keys.

 
 
 
 
 
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Aastra 9112i



Key Features and Benefits
  • Enhanced Call Management — Large storage for personal directory, callers log, and redial list
  • Tight Integration — Support for multiple IP telephony systems including BroadWorks®, Nortel, Sylantro, and Asterisk SIP
  • Remarkable Audio — Quality speakerphone with excellent voice delivery
  • Protect Your Investments — Firmware upgrades can be downloaded and installed in the field as standards develop and protocols evolve

Feature Highlights

  • Personal directory
  • Call forward, Call transfer & Call waiting
  • Caller and calling line information
  • Callers log
  • Conference
  • Redial list
  • Live dial pad or pre-dial support
 
 
Aastra 9133i 
  • Enhanced Call Management: Large storage for personal directory, callers log, and redial list
  • Tight Integration: Support for multiple IP telephony systems including BroadWorks®, Nortel, Sylantro, and Asterisk SIP
  • Multiple Line/Call Appearance Lights: Support up to 9 simultaneous calls. Juggle multiple calls easily with a single press of a button
  • Shared Call Appearances: Shared calls allow you to place a call on hold at one set and pick it up easily at another set
  • Remarkable Audio: Quality speakerphone with excellent voice clarity and delivery
  • Protect Your Investments: Firmware upgrades can be downloaded and installed in the field as standards develop and protocols evolve
  • Less Wiring: Built in dual 10/100 switched Ethernet ports let you share a connection with your computer. Inline power upport eliminates the need for a power adapter
  • AC wall adapter included
  • Compatible with IEEE 802.3af inline power device
  • Optional Power Over Ethernet (POE) injector available for networks that do not supply power
  • Modular headset connector, compatible with amplified business headsets
  • Hearing aid compatible handset
  • Quality speakerphone
  • 3 line adjustable backlit display
  • 3 call appearance lines, 4 navigational keys, 7 programmable keys
  • 10 predefined hard keys including Conference, Call Transfer, Redial, Options, Directory, Save, Delete, Speaker/Headset, Mute
  • Dual 10/100 Mbps switched Ethernet ports
  • Manual or Dynamic Host Configuration Protocol (DHCP) IP address setup
  • Time and date synchronization using SNTP
  • Built-in HTTP server for web administration and maintenance
  • Server provisioned user configuration files
  • Protocols: IETF SIP (RFC3261), G.711 μ-law / A-law, G.729
  • Up to 9 call appearance lines, Personal directory
  • Call forward, Call transfer, Call waiting, Caller and calling line information
  • Callers log, Conference, Redial list
  • Live dial pad or pre-dial support
  • Conferencing, Voicemail, and NAT Traversal support
 
Aastra 53i SIP telephone with PoE

The 53i from Aastra offers powerful features and flexibility in a standards based, carrier-grade basic level IP telephone. With a sleek and elegant design and 3 line LCD display, the 53i is fully interoperable with leading IP Telephony platforms, offering advanced XML capability to access custom applications and support for up to 9 calls simultaneously. Part of the Aastra family of IP telephones, the 53i is ideally suited for light to regular telephone requirements.

  • XML Browser
    The Aastra 53i is equipped with XML browser capabilities allowing access to customized services and applications. Create internal service applications using development guides available from Aastra. This feature provides unlimited potential to customize the Aastra 53i to meet your specific business needs or vertical applications using the display and keypad.
  • Enhanced Call Management
    With extensive storage capacity for personal directories, callers logs and redial lists, the Aastra 53i can improve efficiency by providing more call information with the push of a button. Features such as shared call and bridged line appearances, call forward, call transfer, call waiting, intercom and local 3-way conference increase call flexibility and control.
  • Remarkable Audio
    All Aastra IP telephones offer full-duplex speakerphone with excellent voice clarity and delivery. With years of experience in telecommunications and telephone design, Aastra can ensure superior voice quality on every product we sell.
  • Simplified Deployment
    From initial deployment and configuration to future enhancements and upgrades, the Aastra family of IP telephones are designed to save your business time and money. Dual auto-sensing switched Ethernet ports eliminate additional wiring and simplify installations. Integrated IEEE 802.3af Power-over-Ethernet allows easy deployment with centralized powering and backup. Easily created configuration files, using any text editor, can be used to configure phones individually or centrally.
 
Aastra 55i SIP telephone with PoE

The 55i from Aastra offers powerful features and flexibility in a standards based, carrier-grade advanced level expandable IP telephone. With a sleek and elegant design, 144 x 75 pixel backlit LCD display and 6 dynamic context-sensitive softkeys, the 55i is fully interoperable with leading IP Telephony platforms, offering advanced XML capability to access custom applications and support for up to 9 calls simultaneously.

Part of the Aastra family of IP telephones, the 55i is ideally suited for moderate to heavy telephone users who require more one touch feature keys and XML based programs.

  • XML Browser
    The Aastra 55i is equipped with XML browser capabilities and a large display with dynamic softkeys to easily access customized services and applications. Create internal service applications using development guides available from Aastra. This feature provides unlimited potential to customize the Aastra 55i to meet your specific business needs or vertical applications using the display and keypad.
  • Enhanced Call Management
    With extensive storage capacity for personal directories, callers logs and redial lists and 12 programable keys, the Aastra 55i can improve efficiency by providing more call information with the push of a button. Features such as shared call and bridged line appearances, call forward, call transfer, call waiting, intercom and local 3-way conference increase call flexibility and control.
  • Expandability
    The Aastra 55i supports up to three Aastra 536M modules, each offering 36 keys with LED indicators to create a feature rich attendant console.
  • Remarkable Audio
    All Aastra IP telephones offer full-duplex speakerphone with excellent voice clarity and delivery. With years of experience in telecommunications and telephone design, Aastra can ensure superior voice quality on every product we sell.
  • Simplified Deployment
    From initial deployment and configuration to future enhancements and upgrades, the Aastra family of IP telephones are designed to save your business time and money. Dual auto-sensing switched Ethernet ports eliminate additional wiring and simplify installations. Integrated IEEE 802.3af Power-over-Ethernet allows easy deployment with centralized powering and backup. Easily created configuration files, using any text editor, can be used to configure phones individually or centrally.
 
Aastra 57i SIP telephone with PoE

The 57i from Aastra offers powerful features and flexibility in a standards based, carrier-grade advanced level expandable IP telephone. With a sleek and elegant design, large 144 x 128 pixel graphical backlit LCD display and 6 dynamic context-sensitive softkeys, the 57i is fully interoperable with leading IP Telephony platforms, offering advanced XML capability to access custom applications and support for up to 9 calls simultaneously. Part of the Aastra family of IP telephones, the 57i is ideally suited for executives and heavy telephone users who require more one touch feature keys and a large screen to take full advantage of XML based programs.

  • XML Browser
    The Aastra 57i is equipped with XML browser capabilities and an extra large display with dynamic softkeys to easily access customized services and applications. Create internal service applications using development guides available from Aastra. This feature provides unlimited potential to customize the Aastra 57i to meet your specific business needs or vertical applications using the display and keypad.
  • Enhanced Call Management
    With extensive storage capacity for personal directories, callers logs and redial lists and 12 programable keys, the Aastra 57i can improve efficiency by providing more call information with the push of a button. Features such as shared call and bridged line appearances, call forward, call transfer, call waiting, intercom and local 3-way conference increase call flexibility and control.
  • Expandability
    The Aastra 57i offers two different module extension options. It supports up to three Aastra 536M modules, each offering 36 keys with LED indicators to create a feature rich attendant console. Or, it will also support up to three of the advanced Aastra 560M modules, each offering 60 keys with a screen based LCD display and LED system.
  • Remarkable Audio
    All Aastra IP telephones offer full-duplex speakerphone with excellent voice clarity and delivery. With years of experience in telecommunications and telephone design, Aastra can ensure superior voice quality on every product we sell.
  • Simplified Deployment
    From initial deployment and configuration to future enhancements and upgrades, the Aastra family of IP telephones are designed to save your business time and money. Dual auto-sensing switched Ethernet ports eliminate additional wiring and simplify installations. Integrated IEEE 802.3af Power-over-Ethernet allows easy deployment with centralized powering and backup. Easily created configuration files, using any text editor, can be used to configure phones individually or centrally.
 
Aastra 480i

The Model 480i is an advanced, fully featured, IP screen Telephone that provides a flexible IP solution designed with interoperability and ease of use in mind.

  • Large display — Eight-line backlit screen with six context sensitive softkeys provides more information and flexibility in call handling
  • Enhanced Call Management — Large storage for personal directory, callers log, and redial list
  • Tight Integration — Support for Sphericall MGCP and multiple SIP IP telephony systems including BroadWorks®, Nortel, Sylantro and Asterisk
  • Multiple Line/Call Appearance Lights — Support up to 9 simultaneous calls. Juggle multiple calls easily with a single press of a button
  • Shared Call Appearances — Shared calls allow you to place a call on hold at one set and pick it up easily at another set
  • Remarkable Audio — Quality speakerphone with excellent voice clarity and delivery
  • Protect Your Investments — Firmware upgrades can be downloaded and installed in the field as standards develop and protocols evolve
  • Less Wiring — Built in dual 10/100 switched Ethernet ports let you share a connection with your computer. Inline power support eliminates power adapters
  • Up to 9 call appearance lines
  • Personal directory
  • Call forward
  • Call transfer
  • Call waiting
  • Caller and calling line information
  • Callers log
  • Conference
  • Hold Redial list
  • Redial list
  • Live dial pad or pre-dial support
  • BroadSoft’s BroadWorks enhanced SIP – supports Intercom, Shared Call Appearances, CommPilot Call Manager
  • Nortel’s MCS SIP support – Conferencing, Voicemail, and NAT Traversal support
  • Sphere's Spicall MGCP support - BLF, Intercom
 
536M Expansion Module

The 536M expansion module is designed to increase the power and flexibility of the 5i Series SIP telephones. Up to three modules can be used with the 55i, 57i or 57i CT telephone to create a powerful, feature rich console option. The 536M shares power and signaling with the phone, eliminating the need for additional wiring. Designed for receptionists, administrative assistants, call center agents, power users, and executives who need to monitor and manage a large volume of calls on a regular basis, the 5i Series Expansion Modules provide an intelligent choice for all Enterprise IP environments.

  • Flexible Features
    The 536M is equipped with LED status indicators and supports a variety of programmable features including Line, Speed Dial, Busy Lamp Field, Bridged Line Appearance, Shared Call Appearance and Do Not Disturb.
  • Simplified Deployment
    Sharing both power and signaling with the 5i Series phone, the 536M can be easily added without additional installation or wiring expense. Simplified configuration via WebUI or configuration file from the phone.
  • Expandability
    Up to three 536M modules can be used with the 5i Series models 55i, 57i or 57i CT, offering up to 108 additional keys. Growing businesses can start with a single module and add more as their needs increase.
560M Expansion Module

The 560M expansion module is designed to increase the power and flexibility of the 5i Series SIP telephones. Up to three modules can be used with the 57i or 57i CT telephone to create a powerful, feature rich console option. The 560M shares power and signaling with the phone, eliminating the need for additional wiring. Designed for receptionists, administrative assistants, call center agents, power users, and executives who need to monitor and manage a large volume of calls on a regular basis, the 5i Series Expansion Modules provide an intelligent choice for all Enterprise IP environments.

  • Backlit LCD Screen
    144x128 pixels graphical LCD display with 20 Softkeys with LED’s and 3 hardkeys with LED’s that provide access to three screen levels offering up to a total of 60 programmable keys.
  • Flexible Features
    The 560M is equipped with LED status indicators and supports a variety of programmable features including Line, Speed Dial, Busy Lamp Field, Bridged Line Appearance, Shared Call Appearance and Do Not Disturb.
  • Simplified Deployment
    Sharing both power and signaling with the 5i Series phone, the 560M can be easily added without additional installation or wiring expense. Simplified configuration via WebUI or configuration file from the phone.
  • Expandability
    Up to three 560M modules can be used with the 5i Series models 57i or 57i CT, offering up to 180 additional keys. Growing businesses can start with a single module and add more as their needs increase.

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Дата изменения: 23.03.2016