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Grandstream BudgeTone 201 IP Phone
Recommended for Home / SOHO (Home Office / Small Office) Use
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- Affordable and easy to use/manage single line SIP phone
- Superb audio quality and rich feature functionality
- Single RJ-45 10/100Mbps port
- 3-way conferencing and voicemail indicator with light
- Handsfree speaker phone and downloadable ringtones
- 1 year warranty against hardware defects
- 14 day money back guarantee
- 30 day email support for technical installation issues
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| Please select PSU type required: |
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£37.99
£43.69 inc VAT |
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| Volume Discount: |
5+ |
£36.99
£42.54 inc VAT |
10+ |
£35.99
£41.39 inc VAT |
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| If you need larger quantities or are a reseller then please email us. |
Independent Reviews/Awards
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Build quality is excellent... our top choice for the more serious VoIP subscriber. |
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Editors Choice - With its big buttons, clear easy-to-read display and relatively simple setup, the Grandstream Budgetone is ideal for the home user who just wants a good, basic IP phone. |
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Winner of “Best of Show” at Internet Telephony® Conference & Expo Miami for its open standards, high quality audio, ultra low cost and Grandstream's commitment to excellence and innovation. |
Please note: The reviews/awards listed in the table above are actually for the original BudgeTone 101 which has reached End of LIfe (EOL) and is no longer available.
The Grandstream BudgeTone 201 (BT201) SIP based VoIP Phone offers the same market-leading superb sound quality and rich functionality as the award winning BudgeTone 101. The Budge Tone 201 is ideal for connecting to an Asterisk/Trixbox IP PBX and improves on the BudgeTone 101 as it has supports Secure Real-time Transport Protocol (SRTP), and the speakerphone comes with Advanced Echo Cancellation (AEC) and Acoustic Gain Control (AGC).
Please note: The BT201 provides a single 10/100 RJ-45 10/100Mbps port. The BT200 provides dual RJ-45 10/100Mbps autosensing ports (switched or routed) and an integrated NAT router/DHCP server, which may be more suitable depending on the deployment requirements.
- Certified interoperable with Avaya CM, Cisco CM 5.0 and Broadworks
- 8 Dedicated Feature Keys: Message, Hold, Transfer, Conference, Flash, Speakerphone, Redial/Send, Mute/Del
- Voice Features: Voice Mail Indicator, Redial, Call Log, Volume Control, Caller ID Display or Block, Call Waiting, Hold, Transfer, Forward, FLASH, Mute, 3-Way Conferencing, off-hook auto dial, configurable emergency dialling (e.g., 911), early dial, click-to-dial, auto answer, downloadable ring tones
- Single RJ-45 10/100 autosensing port
- Secure Real-time Transport Protocol (SRTP) and SIP Transport Layer Security (TLS) (pending) support
- Full duplex handsfree speakerphone with Advanced Echo Cancellation (AEC), Acoustic Gain Control (AGC) and side tone support
- Easy configuration through manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment
- NAT-friendly remote firmware upgrade capability via tftp/http even from behind firewalls/NATs
- Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products (e.g., Asterisk®/Trixbox® IP PBX, MS Messenger, Cisco® IP phone and gateway, etc)
- Dynamic negotiation of codec and voice payload length
- SIP and DNS server redundancy and failover
- Features Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
- Advanced and patent pending adaptive jitter buffer control, packet delay and loss concealment technology
- Voice codecs supported include G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.722 (wide-band), G.726 (32K), GSM and iLEB codecs
- Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
- In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
- Support for Fixed IP, DHCP, Automated NAT traversal using IETF STUN (manual configuration of firewall/NAT not required) and symmetric RTP (compatible with Cisco’s ATA-186, etc)
- Protocols supported include SIP RFC3261, TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, PPPoE
- Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
- DIGEST authentication and encryption using MD5 and MD5-sess
- LAN interface: 1 x RJ45 10/100Mbps autosensing
- Headset Jack: 2.5mm Headset port
- LED: 1 red coloured LED
- Phone Case: ABS plastic, 25-button keypad and 12-digit caller ID LCD display
- Universal Switching Power Adaptor: Input: 100-240VAC; 50-60 Hz, Output: +5VDC, 1200mA, UL certified
- Dimensions: 180mm (W) x 222mm (D) x 65mm (H)
- Handset Weight: 690g
- Package Weight: 960g
- Temperature: 32 - 104 ºC (0 - 40 ºF)
- Humidity 10-90% (non condensing)
- Compliance: FCC / CE / C-Tick / RoHS
Asterisk® (www.asterisk.org) is registered trademark of Digium, Inc. Trixbox® (www.trixbox.org) is a registered trademark of Fonality, Inc. Novavox Limited is not affiliated with, nor endorsed by either of the companies listed above.
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