Panasonic KX-UT133 Standard SIP Phone
$79.99
- 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
2 in stock
Description
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