New Improved Yealink TP27 IP Phone – Yealink T27G

Yealink’s new T27G provides users with an upgrade to Gigabit Ethernet and a USB port for optional expansion to features. The Yealink T27G hardware has also been enhanced to give its users a better overall performance.

Yealink T27G IP Phone

What’s new?

  • Gigabit pass through
  • USB2.0 port (pending compatibility with Yealink’s BT40 and WF40 dongles)
  • Opus Codec support for better audio quality
  • Device performance enhancement, faster response on the phones user interface

YEALINK T27G IP Phone

Overview

The T27G has adopted a paperless design, it has 7 programmable memory keys each with an on screen label and dual-colour LED for line status information. There are 3 pages for up to 21 keys in total, for users requiring access to even more feature keys the T27G supports up to 6 EXP20 expansion keypad modules for up to 228 additional keys.

Features

  • 3.66″ (240 x 120 pixels) graphical backlit LCD
  • Dual 10/100/1000 Gigabit Ethernet port
  • PoE support
  • 6 SIP user accounts
  • 7 programmable keys with LED indication (3 page view for a total of 21 keys)
  • Paperless design
  • Hands-free talking (speakerphone)
  • Local directory with 1000 entries
  • Wall mountable (requires wall mount bracket)
  • VPN
  • IPv6 support
  • USB 2.0 port
  • Compatible with up to six EXP20 Expansion Modules
  • Headset support:
    • Corded RJ9 connection
    • Wireless with EHS support (via EHS36)
  • Colour: Black
  • Zero touch secure provisioning
  • Lifetime Warranty

 

DPH500 GSM Desk Phone

Whilst the Westlake® DPH500 GSM Destop Phone is not an IP Telephone it is the ideal desk phone to deploy as an alternative when you don’t have the convenience of VoIP, Ethernet, WiFi or Internet services to deploy an IP Phone solution but need a simple, reliable and easy to operate telephone solution.

DPH500

Simply insert a 2G Network SIM Card into the SIM card slot in the base of the DPH500 GSM Desk Phone, connect the mains power and you now have a mobile phone in the guise of a desk phone.  If you want to make a call, then lift the handset and dial your number (including area code just like you do with a mobile phone) and you can use the DPH500 just like a normal two piece desk phone.

DPH500 GSM Desk Phone

Unattended Reception Phone

You can configure the DPH500 to dial a single number when the handset is lifted – this means that you could leave this GSM Desktop Phone in an unattended reception or location and visitors can simply lift the handset and the phone will call the programmed number.  Whilst in this mode (which is called baby call mode) the DPH500 will not let the user dial any other telephone numbers.

DPH500 For GSM Taxi Phone Installation

Using the Baby Call feature which is also known as Hot Dial, the DPH500 GSM Phone can also be used as a GSM Taxi Phone so could be placed in reception areas of hotels, pubs, clubs or shops so that visitors to those premises could lift the handset and dial the taxi operator to arrange a private hire taxi thus generating extra business for the taxi company that deploys the taxi phone.

Westlake DPH500 is the original version of this popular, low cost GSM Desk Phone – there are many companies that try to imitate the design and features but make sure that you buy the original and best DPH500® GSM Desk Phone – look for the DPH500® Trademark to know you are purchasing from an authorised dealer.

 

 

Cisco SPA303 VoIP/SIP Phone, 3-Line, LCD Display

Cisco SPA303

For small businesses that need affordable, reliable, and easy-to-use IP phones, the Cisco SPA303 3-Line IP Phone with Ethernet Ports and LCD display can be used with either Session Initiation Protocol (SIP) or Smart Phone Control Protocol (SPCP) and is supported with the Cisco Unified Communications 500 Series, Cisco SPA9000 Voice System, and hosted IP telephony systems. The SPA303 enables many productivity-enhancing features and XML applications. It is easy to install and supports highly secure remote provisioning as well as web-based configuration.

spa303

Key Features

  • 3-line business-class IP phone.
  • Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX).
  • Dual switched Ethernet ports, speakerphone, caller ID, call hold, conferencing, and more.
  • Easy installation and highly secure remote provisioning, as well as menu-based and web-based configuration.
  • Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol

Specification

Telephony Features
  • Three voice lines
  • Pixel-based display: 128 x 64 monochrome graphical liquid crystal display (LCD)
  • Line status: active line indication, name and number
  • Menu-driven user interface
  • Shared line appearance*
  • Speakerphone
  • Call hold
  • Music on hold*
  • Call waiting
  • Caller ID name and number
  • Outbound caller ID blocking
  • Call transfer: attended and blind
  • Three-way call conferencing with local mixing
  • Multiparty conferencing via external conference bridge
  • Automatic redial of last calling and last called numbers
  • On-hook dialling
  • Call pickup: selective and group*
  • Call park and unpark*
  • Call swap
  • Call back on busy**
  • Call blocking: anonymous and selective
  • Call forwarding: unconditional, no answer, and on busy
  • Hot line and warm line automatic calling
  • Call logs (60 entries each): made, answered, and missed calls
  • Redial from call logs
  • Personal directory with auto-dial (100 entries)
  • Do not disturb
  • Digits dialled with number auto-completion
  • Anonymous caller blocking
  • Support for Uniform Resource Identifier (URI) (IP) dialling (vanity numbers)
  • On-hook default audio configuration (speakerphone and headset)
  • Multiple ring tones with selectable ring tone per line
  • Called number with directory name matching
  • Ability to call number using name: directory matching or via caller ID
  • Subsequent incoming calls show calling name and number
  • Date and time with support for intelligent daylight savings
  • Call duration and start time stored in call logs
  • Call timer
  • Name and identity (text) displayed at start-up
  • Distinctive ringing based on calling and called number
  • 10 user-downloadable ring tones
  • Speed dialling, eight entries
  • Configurable dial/numbering plan support
  • Intercom*
  • Group paging
  • Network Address Translation (NAT) traversal, including Serial Tunnel (STUN) support
  • DNS SRV and multiple A records for proxy lookup and proxy redundancy
  • Syslog, debug, report generation, and event logging
  • Support for highly secure encrypted voice communications
  • Built-in web server for administration and configuration with multiple security levels
  • Automated remote provisioning, multiple methods; up to 256-bit encryption (HTTP, HTTPS, Trivial File Transfer protocol [TFTP])
  • Option to require administrator password to reset unit to factory defaults

 

Hardware Features

 

  • Pixel-based display: 128 x 64 monochrome LCD graphical display
  • Dedicated illuminated buttons for:
    • Audio mute on/off
    • Headset on/off
    • Speakerphone on/off
  • Four-way rocking directional knob for menu navigation
  • Voicemail message waiting indicator light
  • Voicemail message retrieval button
  • Dedicated hold button
  • Settings button for access to feature, setup, and configuration menus
  • Volume control rocking up/down knob controls handset, headset, speaker, ringer
  • Standard 12-button dialling pad
  • High-quality handset and cradle
  • Built-in high-quality microphone and speaker
  • Headset jack: 2.5 mm
  • LED test function
  • Two Ethernet LAN ports with integrated Ethernet switch: 10/100BASE-T RJ-45
  • 5 VDC universal (100-240V) switching included
Security Features

 

  • Password-protected system, preset to factory defaults
  • Password-protected access to administrator and user-level features
  • HTTPS with factory-installed client certificate
  • HTTP digest: encrypted authentication via MD5 (RFC 1321)
  • Up to 256-bit Advanced Encryption Standard (AES) encryption

 

Data Networking

 

  • MAC address (IEEE 802.3)
  • IPv4 (RFC 791)
  • Address Resolution Protocol (ARP)
  • DNS: A record (RFC 1706), SRV record (RFC 2782)
  • Dynamic Host Configuration Protocol (DHCP) client (RFC 2131)
  • Internet Control Message Protocol (ICMP) (RFC 792)
  • TCP (RFC 793)
  • User Datagram Protocol UDP (RFC 768)
  • Real Time Protocol RTP (RFC 1889, 1890)
  • Real Time Control Protocol (RTCP) (RFC 1889)
  • Real Time Control Protocol – Extended Report ( RTCP-XR) ( RFC 3611 )
  • Differentiated Services (DiffServ) (RFC 2475)
  • Type of service (ToS) (RFC 791, 1349)
  • VLAN tagging 802.1p/Q: Layer 2 quality of service (QoS)
  • Simple Network Time Protocol (SNTP) (RFC 2030)

 

Dimensions (W x H x D)
  • 8.66 x 7.80. x 1.18 in. (220 x 198 x 30 mm)
Voice Gateway

 

  • SIP version 2 (RFC 3261, 3262, 3263, 3264)
  • SPCP with the Cisco Unified Communications 500 Series
  • SIP proxy redundancy: dynamic via DNS SRV, A records
  • Re-registration with primary SIP proxy server
  • SIP support in NAT networks (including STUN)
  • SIPFrag (RFC 3420)
  • Highly secure (encrypted) calling via Secure Real-Time Transport Protocol (SRTP)
  • SIP/TLS
  • Codec name assignment
  • Voice algorithms:
    • G.711 (A-law and µ-law)
    • G.726 (16/24/32/40 kbps)
    • G.729 AB
    • G.722
  • Dynamic payload support
  • Adjustable audio frames per packet
  • Dual-tone multifrequency (DTMF), in-band and out-of-band (RFC 2833) (SIP INFO)
  • Flexible dial plan support with interdigit timers
  • IP address/URI dialling support
  • Call progress tone generation
  • Jitter buffer: adaptive
  • Frame loss concealment
  • Voice activity detection (VAD) with silence suppression
  • Attenuation/gain adjustments
  • Message waiting indicator (MWI) tones
  • Voicemail waiting indicator (VMWI), via NOTIFY, SUBSCRIBE
  • Caller ID support (name and number)
  • Third-party call control (RFC 3725)

Yealink T46GN VoIP/SIP Phone, 16-Line w/ PoE

Yealink T46GN

The Yealink T46GN is the latest revolutionary VoIP/SIP Phone for executive users and busy professionals, with a high-resolution TFT colour display to deliver a rich visual experience and Yealink Optima HD technology for rich, clear, life-like voice communications. This Yealink T46GN supports Gigabit Ethernet, a variety of device connections, including EHS headset and USB, and with its programmable keys, it supports vast productivity-enhancing features.

Key Features

  • Revolutionarily new design with Yealink Optima HD voice.
  • Dual-port 10/100/1000 RJ45 Gigabit Ethernet.
  • 4.3″ 480 x 272-pixel colour display with backlight.
  • Built-in a USB port, support Bluetooth headset (Through USB Dongle).
  • Power over Ethernet PoE support.
  • Headset, EHS support.
  • Supports expansion modules.
  • Stand with 2 adjustable angles.
  • Simple, flexible and secure provisioning options.

Cisco SPA504G

Specifications

Audio Features

 

  • HD voice: HD handset, HD speaker
  • Codecs: G.722, G.711(A/µ), G.723, G.729AB, G.726, iLBC
  • DTMF: In-band, Out-of-band (RFC 2833) and SIP INFO
  • Full-duplex hands-free speakerphone with AEC
  • VAD, CNG, AEC, PLC, AJB, AGC

 

Phone Features

 

  • 16 VoIP accounts
  • Call hold, mute, DND
  • One-touch speed dial, hotline
  • Call forward, call waiting, call transfer
  • Group listening, SMS, emergency call
  • Redial, call return, auto answer
  • 3-way conferencing
  • Direct IP call without SIP proxy
  • Ring tone selection/import/delete
  • Set date time manually or automatically
  • Dial plan
  • XML Browser
  • Action URL/URI
  • RTCP-XR (RFC3611), VQ-RTCPXR (RFC6035)
Directory

 

  • Local phonebook up to 1000 entries
  • Black list
  • XML/LDAP remote phonebook
  • Intelligent search method
  • Phonebook search/import/export
  • Call history: dialled/received/missed/forwarded
IP-PBX Features

 

  • Busy Lamp Field (BLF)
  • Bridged Line Appearance(BLA)
  • Anonymous call, anonymous call rejection
  • Hot-desking
  • Message Waiting Indicator (MWI)
  • Voice mail
  • Call park, call pickup
  • Intercom, paging
  • Music on hold
  • Call completion
  • Call recording
Display and Indicator

 

  • 4.3″ 480 x 272-pixel colour display with backlight
  • 16-bit depth colour
  • LED for call and message waiting indication
  • Dual-colour (red or green) illuminated LEDs for line status information
  • Wallpaper
  • Intuitive user interface with icons and soft keys
  • National language selection
  • Caller ID with name, number and photo

 

Feature keys

 

  • 10 line keys with LED
  • 10 line keys can be programmed up to 27 various features (3-page view)
  • 7 features keys: message, headset, hold, mute, transfer, redial, hands-free speakerphone
  • 4 context-sensitive “soft” keys
  • 6 navigation keys
  • Volume control keys
  • Illuminated mute key
  • Illuminated headset key
  • Illuminated hands-free speakerphone key
Other Physical Features

 

  • Stand with 2 adjustable angles
  • Wall mountable
  • External universal AC adapter (optional): AC 100~240V input and DC 5V/2A output
  • Power consumption (PSU): 1.8-5.4W
  • Power consumption (PoE): 2.1-8.0W
  • Dimension(W*D*H*T): 244mm*213mm*185mm*54mm
  • Operating humidity: 10~95%
  • Operating temperature: -10~50C
Management

 

  • Configuration: browser/phone/auto-provision
  • Auto provision via FTP/TFTP/HTTP/HTTPS for mass deploy
  • Auto-provision with PnP
  • BroadSoft device management
  • Zero-sp-touch, TR-069
  • Phone lock for personal privacy protection
  • Reset to factory, reboot
  • Package tracing export, system log

 

Network and Security

 

  • SIP v1 (RFC2543), v2 (RFC3261)
  • Call server redundancy supported
  • NAT transverse: STUN mode
  • Proxy mode and peer-to-peer SIP link mode
  • IP assignment: static/DHCP/PPPoE
  • HTTP/HTTPS web server
  • Time and date synchronization using SNTP
  • UDP/TCP/DNS-SRV(RFC 3263)
  • QoS: 802.1p/Q tagging (VLAN), Layer 3 ToS DSCP
  • SRTP for voice
  • Transport Layer Security (TLS)
  • HTTPS certificate manager
  • AES encryption for configuration file
  • Digest authentication using MD5/MD5-sess
  • OpenVPN, IEEE802.1X
  • IPv6

 

Cisco SPA504G IP Phone, 4 Lines, 2 Ethernet Ports, PoE

Cisco SPA504G

The Cisco SPA504G IP Phone is designed to improve and simplify communications across your entire company. From the office to the meeting room, a Cisco SPA504G IP Phone offers enough features for everyone from executives and office workers to staff on your manufacturing floor. Your employees stay productive, and enjoy reliable access to voice and data communications wherever they go, while your costs stay low.

Key Features

  • Full-featured business-class IP phone supporting Power over Ethernet (PoE)
  • Monochrome backlit display for ease of use, aesthetics, and onscreen applications
  • Connects directly to a hosted IP telephony service or an IP private branch exchange (PBX)
  • Wideband audio for unsurpassed voice clarity and enhanced speaker quality
  • Easy installation and highly secure remote provisioning, as well as menu-based and web-based configuration
  • Supports up to two Cisco SPA500S 32-Button Attendant Consoles, adding up to 64 additional buttons
  • Supports both Session Initiation Protocol (SIP) for Cisco SPA9000, open source, and hosted telephony solutions and Smart Phone Control Protocol (SPCP) for Cisco Unified Communications 500 Series for Small Business
  • Play back and view personal MP3 files and photos through on-phone application

Cisco SPA504G

Specifications

Features ·         Four voice lines

·         Four Independent SIP Registrations*

·         Line status: active line indication, with name and number

·         Menu-driven user interface

·         Shared line appearance**

·         Speakerphone

·         Call hold

·         Music on hold**

·         Call waiting

·         Caller ID name and number

·         Outbound caller ID blocking

·         Call transfer: attended and blind

·         Three-way call conferencing with local mixing

·         Multiparty conferencing via external conference bridge

·         Automatic redial of last calling and last called numbers

·         On-hook dialling

·         Call pickup: selective and group**

·         Call park and unpark**

·         Call swap

·         Call back on busy

·         Call blocking: anonymous and selective

·         Call forwarding: unconditional, no answer, on busy

·         Hot line and warm line automatic calling

·         Call logs (60 entries each): made, answered, and missed calls

·         Redial from call logs

·         Personal directory with auto-dial (100 entries)

·         Digits dialled with number auto-completion

·         Anonymous caller blocking

·         Uniform Resource Identifier (URI) (IP) dialling support (vanity numbers)

·         On-hook default audio configuration (speakerphone and headset)

·         Multiple ring tones with selectable ring tone per line

·         Called number with directory name matching

·         Ability to call number using name: directory matching or via caller ID

·         Subsequent incoming calls show calling name and number

·         Date and time with support for intelligent daylight savings

·         Call start time stored in call logs

·         Call timer

·         Name and identity (text) displayed at start-up

·         Distinctive ringing based on calling and called number

·         10 user-downloadable ring tones

·         Speed dialling, eight entries

·         Configurable dial/numbering plan support

·         Group paging

·         Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs (STUN) support

·         DNS SRV and multiple A records for proxy lookup and proxy redundancy

·         Syslog, debug, report generation, and event logging

·         Highly secure call encrypted voice communications support

·         Built-in web server for administration and configuration with multiple security levels

·         Automated remote provisioning, multiple methods; up to 256-bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP])

·         Option to require administrator password to reset unit to factory defaults

 

Hardware Features ·         Pixel-based display: 128 x 64 monochrome LCD graphical display with backlight

·         Dedicated illuminated buttons for:

–          Audio mute on/off

–          Headset on/off

–          Speakerphone on/off

·         4-way rocking directional knob for menu navigation

·         Voicemail message waiting indicator (VMWI) light

·         Voicemail message retrieval button

·         Dedicated hold button

·         Settings button for access to feature, setup, and configuration menus

·         Volume control rocking up/down knob controls handset, headset, speaker, ringer

·         Standard 12-button dialing pad

·         High-quality handset and cradle

·         Built-in high-quality microphone and speaker

·         Headset jack: 25 mm

·         LED test function

·         Two Ethernet ports with integrated Ethernet switch: 10/100BASE-T RJ-45

·         802.3af compliant PoE

·         Optional 5 VDC universal (100-240V) switching

 

Regulatory Compliance ·         FCC (Part 15, Class B), CE Mark, A-Tick, C-Tick, Telepermit, UL, CB
Security Features

 

·         Password-protected system, preset to factory default

·         Password-protected access to administrator and user-level features

·         HTTPS with factory-installed client certificate

·         HTTP digest: encrypted authentication via MD5 (RFC 1321)

·         Up to 256-bit Advanced Encryption Standard (AES) encryption

·         SIP over Transport Layer Security (TLS)

·         Secure Real-Time Transport Protocol (SRTP)

Yealink T42G VoIP-SIP telephone, 3-Line w/ PoE

 Yealink T42G

The Yealink T42G is an entry level and feature-rich sip phone for business, and with 3-Lines, this IP Phone has been designed to deliver a superb sound quality as well as rich visual experience. This Yealink T42G IP Phone supports seamless migration to GigE-based network infrastructure, and with programmable Keys it supports vast productivity-enhancing features. This handset also uses standard encryption protocols to perform highly secure remote provisioning and software upgrades.

Yealink T42G

Key Features

  • Revolutionarily new design with Optima HD voice
  • Dual-port 10/100/1000 Gigabit Ethernet
  • 2.7″ 192×64-pixel graphical LCD with backlight
  • Up to 3 SIP accounts
  • Power over Ethernet (PoE) support
  • Headset, EHS support
  • Integrated stand with 2 adjustable angles
  • Simple, flexible and secure provisioning options

Specifications

Audio Features

 

  • HD voice: HD handset, HD speaker
  • Wideband codec: G.722
  • Narrowband codec: G.711(A/µ), G.723, G.729AB, G.726
  • DTMF: In-band, Out-of-band (RFC 2833) and SIP INFO
  • Full-duplex hands-free speakerphone with AEC
  • VAD, CNG, AEC, PLC, AJB, AGC
Phone Features

 

  • 3 VoIP accounts
  • One-touch speed dial, redial, call forward, Call waiting, Call transfer, Call hold, Call return, Group listening, Mute, Auto answer, DND, SMS
  • Call history: dialled/received/missed/forwarded
  • 3-way conference call
  • Direct IP call without SIP proxy
  • Ring tone selection/import/delete
  • Hotline, Emergency call
  • Set date time manually or automatically
  • Dial Plan, XML Browser, Action URL
Display and Indicator

 

  • 2.8” 192×64-pixel 4-level greyscale graphical
  • LCD with backlight
  • LED for call and message waiting indication
  • Dual-colour (red or green) illuminated LEDs for line status information
  • Intuitive user interface with icons and soft keys
  • National language selection
  • Caller ID with name, number
Feature Keys

 

  • 6 line keys with LED
  • 6 line keys can be programmed up to 15 various features (3-page view)
  • 5 features keys: Message, Headset, Mute, Redial, Hands-free speakerphone
  • 4 context-sensitive “soft” keys
  • 6 navigation keys
  • 2 volume control keys
  • Illuminated mute key
  • Illuminated headset key
  • Illuminated hands-free speakerphone
Other Physical Features

 

  • Stand with 2 adjustable angles
  • Wall mountable
  • Power adapter: AC 100~240V input and DC 5V/ 1.2A output
  • Dimensions (W x D x H x T): 212mm x 189mm x 175mm x 54mm
  • Operating humidity: 10~95%
  • Storage temperature: -10~50°C
Network and Security

 

  • SIP v1 (RFC2543), v2 (RFC3261)
  • NAT transverse: STUN mode
  • Proxy mode and peer-to-peer SIP link mode
  • IP assignment: static/DHCP/PPPoE
  • HTTP/HTTPS web server

Cisco SPA502G 1-Line IP Phone with Display, PoE and PC Port

Cisco SPA502G

The Cisco SPA502G IP Phone is designed to improve and simplify communications across your entire company. From the office to the meeting room, a Cisco SPA502G IP Phone offers enough features for everyone from executives and office workers to staff on your manufacturing floor. Your employees stay productive, and enjoy reliable access to voice and data communications wherever they go, while your costs stay low.

Integrated communications on the Cisco SPA502G IP Phone lets employees make voice calls and access corporate data, such as company directories, quickly and easily, no matter where they happen to be working.

Employees can take advantage of their Cisco SPA502G phone systems features, including: speakerphone, redial, call transfer, conferencing, paging, intercom, volume control, message-waiting and voicemail indicator lights, mute key, headset use, and call history directories.

Cisco SPA502G

Features:

  • 1-Line Phone
  • Dual switched ethernet ports
  • IEEE 802.3af PoE (Power Over Ethernet)
  • Supports SIP or SPCP protocol
  • G.729 and G.722 Wideband Audio Support
  • Menu driven user interface
  • Web based GUI, remote provisioning
  • IP dialing support
  • Hi Res LCD Display
  • 2.5mm stereo headset connector
  • AUX port (SPA932 or SPA500S attendant console is supported)
  • High quality audio handset
  • Full duplex hands free speakerphone
  • Hearing aid compatibility (HAC)
  • Support calls between TTY’s
  • Works well with GN Netcom, Jabra and Plantronics wired headsets
  • Optional PA100 Power Adapter and MB100 Wall Mount Adapter

Specifications:

Product Type 
Main Features

 

Multiple VoIP protocol support, integrated Ethernet switch, Power over Ethernet (PoE) support
VoIP Protocols SIP, SIP v2, SPCP
Voice Codecs   G.726 G.722, G.729a, G.711u, G.711a,
Lines Supported Single-line
Intercom Yes
Speakerphone Yes (digital duplex)
Caller ID Yes
Voice Mail Capability Yes
Automatic Redial Yes
Display LCD display monochrome
Display Features Backlit
Network Ports Qty.. 2 x Ethernet 10Base-T/100Base-TX
Body Colour Silver, dark grey
General
Product Type VoIP phone
Body Color Silver, dark grey
Body Material ABS plastic
Phone Features
Key Expansion Module Max Qty 2
Dialer Type Keypad
Dialer Location Base
Conference Call Capability 3-way
Intercom Yes
Speakerphone Yes (digital duplex)
Caller ID Yes
Automatic Redial Yes
Voice Mail Capability Yes
Call Waiting Yes
Call Forwarding Yes
Call Transfer Yes
Call Hold Yes
Menu Operation Yes
Function Buttons Conference button, speakerphone button, transfer button, headset button, mute button, hold button, redial button
Volume Control Yes
Ringer Control Yes
Indicators Voice message waiting indicator, speakerphone indicator, headset
Firmware Upgradable Yes
Additional Functions Call timer, built-in clock, pager call
Additional Features On-hook dialling, music on hold, built-in web server
IP Telephony
Main Features Multiple VoIP protocol support, integrated Ethernet switch, Power over Ethernet (PoE) support
VoIP Protocols SIP, SIP v2, SPCP
Voice Codecs G.722, G.729a, G.711u, G.711a, G.726
Lines Supported Single-line
Quality of Service IEEE 802.1Q (VLAN), Differentiated Services (DiffServ), IEEE 802.1p, Type of Service (ToS)
IP Address Assignment DHCP
Security 256 bit AES
Network Protocols IP, TCP, TFTP, UDP, ICMP, ARP, HTTP, DNS, HTTPS
Network Ports Qty 2 x Ethernet 10Base-T/100Base-TX
Voice Features Comfort noise generation (CNG), voice activity detection (VAD), HD Voice
Network Features Network Address Translation (NAT)
Phone Memory
Phone Directory Capacity 100 names & numbers
Call Log Capacity 60 numbers
Display
Type LCD display – monochrome
Display Location Base
Display Resolution 128 x 64 pixels
Features Backlit
Miscellaneous
Connections 1 x headset jack / sub-mini-phone 2.5 mm
Cables Included Network cable
Compliant Standards CE, UL, C-Tick, CB, FCC Part 15 B, A-Tick
Dimensions & Weight (Base)
Width 8.4 in
Depth 1.7 in
Height 8.3 in
Weight 2 lbs

Aastra 6725 IP Desk Phone

Aastra 6725 IP

The Aastra 6725 IP phone is optimized for Microsoft Communicator. The Aastra 6725ip supports a Gigabit Ethernet interface which when connected to Microsoft Communications Server “14”, it becomes a powerful Unified Communications (UC) device. This new Aastra phone enjoys the same exceptional voice quality and proven Aastra reliability in a stylish global design.

Aastra 6725ip

Features:

  • 3.5″ QVGA Color LCD Screen
  • 2-way Navigation Key and Select Key
  • 2 LCD Softkeys and Menu Key
  • Home and Back Keys
  • Wideband Audio Handset
  • Wideband speakerphone
  • 2 x 10/100/1000 Ethernet Ports
  • 1 USB Type-A
  • 1 USB Type-B
  • UC Presence Indicator
  • Message Waiting indicator
  • PoE Enabled
  • Optional AC Adapter

Benefits:

 “Always On” Operation

 It is no longer necessary to have a phone connected to an operating PC to enjoy the benefits of Lync. Both models of Aastra’s Microsoft Lync phones are true standalone IP phones and have the Microsoft Lync 2010 Phone Edition embedded in the phone. The Aastra 6725ip has the additional advantage of being able to be connected to a local desktop PC via a USB cable, enabling desktop PC phone control, multimedia call escalation and synchronized PC and phone locking.

Intuitive Interface

An ergonomic and user friendly hard key layout, used in conjunction with selectable soft keys, a large 2-way navigation key and a 3.5″ LCD display screen, provide an efficient and intuitive interface.

UC Presence Indicator

The Aastra 6725ip includes a large presence indicator displaying, at a glance, your own presence status being shown to other Lync users.

Simplified Deployments

Both the Aastra 6721ip and Aastra 6275ip are designed to save your business time and money. Dual autosensing switched Gigabit Ethernet ports eliminate additional wiring and simplify installations without compromising the bandwidth required by “power users”. Integrated IEEE 802.3af Power Over Ethernet allows easy deployment with centralized powering and backup.

Cisco Unified IP Phone 7940G

Cisco 7940G

The Cisco 7940G is dynamic and designed to grow with system capabilities. Features will keep pace with new changes via software updates to the phone’s flash memory. The 7940G includes a large pixel based LCD display. The graphic capability of the display allows for the inclusion of such features as XML (Extensible Markup Language) and future features. The 7940G is also compatible with Cisco power over Ethernet (PoE) technology which eases installation and reduces clutter in the office. It features an adjustable foot stand that can be changed from flat to 60 degrees.

The Cisco IP Phone 7940G, a key offering in the IP Phone portfolio, addresses the communication needs of a transaction type worker. It provides two programmable line and feature keys, plus a high-quality speakerphone. The Cisco IP Phone 7940G also has four dynamic soft keys that guide users through call features and functions. Built-in headset port and integrated Ethernet Switch are standard with the Cisco IP Phone 7940G. Also, includes audio controls for full duplex speakerphone, handset and headset. The Cisco IP Phone 7940G also features a large, pixel-based LCD display. The display provides features such as date and time, calling party name, calling party number, and digits dialed.

cisco cp7940G

Features:

  • 2 programmable line and feature buttons
  • 24+ user-adjustable ringtones
  • Comfort noise generation and voice activity detection (VAD) programming on a system basis
  • Adjustable display contrast
  • Integrated headset jack
  • The graphic capability of the display allows for the inclusion of such features as XML (Extensible Markup Language) and future features.

The Cisco 7940G is a high end office worker or manager phone. It’s superb sound quality and features make the 7940G arguably the most popular IP phone in the market. The 7940G is a two line phone with a full duplex (two way audio) speakerphone and a dedicated headset port (RJ11). It is designed for medium to high call volume and is a great phone for a cubicle or office worker.

 

Model CP-7940G
# of Lines 2
Backlit Display No
Colour Display No
Touchscreen Display No
Gigabit No
Headset Interface RJ9
EHS Support No

Panasonic KX-UT133 SIP Phone

KX-UT133

 

The Panasonic UT range of SIP telephony terminals enhance personal communications through excellent HD quality audio on every phone, combined with easy access to powerful supportive features and applications.

The range, from standard phones, office key-sets, executive terminals and touch-screen Smart Desk application phones, addresses all requirements. Panasonic’s reputation for design, quality, reliability and care for the environment, ensures an exceptional user experience wherever the terminals are deployed – as part of a “cloud based” service or with an IP PBX – in a business environment or in the home.

panasonic_kx-ut133

Features:

  • Certified for Asterisk and Broadsoft
  • 24 Feature keys
  • 3-way conference call
  • XML Application Interface
  • 500 Entry Phonebook
  • 3 Line backlit LCD Display
  • Electric hook switch (Plantronics)
  • 2 Ethernet ports, PoE
  • Green (Low standby power consumption)

 

High Definition “HD” Audio

The KX-UT series of SIP terminals offers ‘best in class’ audio quality, meaning fewer repeated conversations and misheard calls. Offering Wideband High Definition Audio as standard across the range, the UT series offer G.722, G.711 , G.726 and G.729a Codecs. Coupled with Enhanced Echo Cancellation and an Expanded Acoustic Chamber, the UT series of SIP terminals offers a superior audio experience to users over handset, speakerphone and optional headsets. The entire range features wideband compliant, hearing aid compatible handsets, and built-in, high quality speaker and microphone.

LCD Displays

Large, clear LCD displays with intuitive User Interface offer fast access to phonebooks and features.

Electric Hook Switch

A built in Electronic Hook Switch (Plantronics compliant) port allows the KX-UT133 and KX-UT136 SIP Terminals to have access to the Plantronics range of DECT enabled headsets. This offers a range of portability and comfort as frequent users are able to move around freely, without being tied down by handsets.

ECO Friendly

Low power consumption, combined with an advanced ECO standby mode means lower energy costs. Consumption can be less than 1 watt in ECO mode! (UT 1xx series).

Plug and Play Configuration

The entire KX-UT range of SIP devices features support for extensive provisioning options which allow automatic, quick device configuration, using configuration files stored on a remote server, meaning a smaller administration overhead, saving time and money. The range is certified compatible with Digium Asterisk and Broadsoft Broadworks, ensuring excellent compatibility with leading soft switch suppliers.

Reduced cabling – One wire to the desktop

Reduce the need to rewire the office when you expand your business. Many of the Panasonic SIP terminals include a second network port, allowing a second device to access the network with less cabling. This reduction extends to the power supply – all the UT series terminals support Power Over Ethernet. No more power adaptors cluttering up desks.

Integration with CRM applications

Connecting the Panasonic UT series terminals to a Broadsoft hosted service opens a world of CRM integration. Both incoming and outgoing calls can be handled from the desktop. By using Mondago Go Connect, access to a wide range of commonly used CRM systems can be achieved. Desktop and web based applications can now be used, offering productivity and a competitive edge.