+1 858 578-0404

Panasonic KX-UT133 Standard SIP Phone


Standard SIP Phone with 3 line Backlit LCD Display
  • HD Voice and PoE Enabled
  • Two Data Ports* for Simplified Wiring and Connecting a Second Device
  • Certified for Use with Broadsoft Broadworks and Digium® Asterisk®
  • Environmentally Friendly – Power Consumption 1W in ECO Model
Flexible, Cost-Effective Communications Solutions

2 in stock

SKU: UT133-B Category: Tags: ,


Panasonic KX-UT133 Standard SIP Phone Enhance your business with the latest SIP telephone technology, helping to improve your business communications, decrease running costs and simplify equipment management. The KX-UT133 SIP telephone is ideal for all office use, suitable for reception, or hotel rooms.

Panasonic KX-UT133 Standard SIP Phone 24 keys enable traditional “key-set” working, improved familiarity, reduced training; instant feature access. This SIP device has a rich feature set, including a clear 3 line alphanumeric display, caller ID, call log, 3-way conference communication, and many more provided by your IP-PBX*, Asterisk, or Broadsoft service provider. KEY FEATURES

  • Ideal for all business and home office use, reception areas, hotel room
  • High quality “HD Audio” device
  • Full-duplex speakerphone
  • Dual Network port (connection for a PC)
  • 3 line Backlit display, PoE, and XML sup
  • LCD Display: Monochrome Graphical
  • LCD Size: 242 x 55 pixels – 3 lines
  • LCD Contrast: 6 levels
  • LCD Backlight: On/Auto/Off
  • Desk mount tilt: No
  • Wall mount: KX-A432-B (optional)
  • Power adapter: KX-A239 (optional)
  • Ethernet Ports: 2 – 10/100
  • Power over Ethernet (PoE): Yes (Class 2)
  • Audio Codec: G.711, G.722, G.726, G.729a
  • Handset, Speaker, Headset Volume: 8 levels (includes echo cancellation and distortion prevention) 15 levels
  • Ringtones: 27
  • Ringer Volume: 6 levels + Off
  • Headset Port: 2.5 mm
  • Hearing Aid: Compatible with Hearing Aid TIA-801
  • Keys (total): 28
  • FF Keys: –
  • Navigator and Cancel Key: Yes
  • Phone Book (Entries): 500 – each with 5 numbers
  • Call Log Entries: 30 incoming calls + 30 outgoing calls
  • Conferencing: 3 parties (within terminal – multi-party dependent on server)
  • SIP Accounts: 2
  • SIP Compatibility: RFC 3261 Standard SIP Server, Asterisk, Broadsoft
  • IP Version: IPv4
  • DHCP Client: Yes
  • DNS: Yes
  • HTTP / HTTPS: Yes
  • SNTP Client: Yes
  • VLAN (802.1q): Yes
  • QoS (DiffServ): Yes
  • Plug & Play Configuration: Server based configuration, TR-069, BroadSoft Device Management Server
  • Manual Configuration: Internal web Configurator, Local (LCD based) network configuration