Artdio IPF-2000

Artdio IPF-2000 (IPF2000)

IPF-2000
Artdio VoIP IP Phone SIP H323 Asterisk IPF-2000 101 102
 

 

Artdio VoIP Phone
Model# IPF-2000

Compatible

ArtDio IPF-2000 series VoIP phone is designed to be easily deployed with new service to existing infrastructure of small and medium size enterprises and service providers while minimizing overall operating costs.

Cost-Effective
ArtDio IPF-2000 series VoIP phone can save you monthly phone bills and reduce operating costs by converging voice and data. Its free charge to make phone long-distance calls by Internet. Or you can setup a DID number in IPF-2000 that guarantees savings on monthly bills and while keeping your VoIP phone as a common phone. No need to learn dialing procedure, just pick up the phone and dial.

Minimum Infrastructure Requirements
To program the IPF-2000 series phone, you can use any web browser. IPF-2000 series VoIP phone can be operated and used without a computer, what you need is Broadband Internet Connection and a common router and presto - you are ready to go!

High Quality Phone Call
Featuring the adaptive jitter buffer that adjusts to changes in bandwidth, delay and background noise, IPF-2000 series VoIP phone offers excellent voice quality.

Compatibility
IPF-2000 is fully compatible with H.323 v4, MGCP and SIP standards, and supports new broadband solutions, such as xDSL, Cable Modem.

ArtDio IPF-2000 is a full-featured VoIP telephone with hands-free speaker phone. It is best suited for residential users, SOHO, and small businesses. IPF-2000 is interoperable with the most used protocols in the internet telephony industry. Users can easily register with a service provider that offers SIP, H.323 or MGCP services and benefit from their cost saving communication plans.

With ArtDios unique VoIP technology, IPF-2000 is the leader in voice quality amongst competitors. Its plug-and-play capability allows internet telephony service providers to easily deploy IPF-2000 at their clients premises.

Voice
Call Control ProtocolGH.323 V4 or MGCP or SIP(option)
Voice CompressionGG.711 A/g -Law, G.723.1, G.729A/B/AB
Support Silence Suppression, VAD, CNG, Acoustic Echo Cancellation, Jitter Buffering
Echo CancellationGG.165 16ms
Delayed (End to End)G< 100ms
Flow of the averageG10-12k bit/s
DTMF tone detection
E.164 Dial plan

LAN Interfaces
Interface: 10 Base-T Ethernet
Connectors Type: RJ-45 connector

Networking Protocol
PPPoE
DHCP client/Static IP
UDP/TCP
RTP/RTCP
FTP/HTTP/DNS

Dialing Methods
On-hook dialing
Redial key dialing the last dialed numbers
Extension number dialing through gatekeeper/SIP proxy server
Direct IP address dialing to IP phone or gatekeeper/ SIP proxy server

Management Functions
Web browser based management provided
Phone set configuration
DisplayG32 characters LCD display(2 lines with 16 characters each)

Environmental
Operation TemperatureG0 - 40J (32X - 104XF)
Storage TemperatureG-30 - 65J (-22X - 149XF)
Relative HumidityG10 V 95% Non-Condensing

Power
Input AC RangeG100 V 240VAC, 50 V 60Hz
Input DC RangeG5VDC, 1A
Power ConsumptionG3W

Other Information
Data MemoryG16MB
Procedure MemoryG8MB Flash memory
NAT Pass-through(SIP with STUN)
Call Hold, Call Mute, Call Status, Caller ID
Headset Connector, Speaker Phone, Networking Status
Phone Book, Volume Adjustment, Firmware Upgrade
No-Answer Call Forward, Busy Call Forward, Always Call Forward
Miss-Call, Dialed-Call, Answered-Call, Last Number Redial, Speedy Dial
Voice Response, Web configuration, LCD Display, LED indicators...etc.
Feature KeysGCalled, Answered, Miss-Call, Phone, Network-Setting(IP/Subnet Mask/Gateway/Server IP)
Speedy-Dial, Call-Hold, Redial, Mute, Volume Up/Down, Handsfree

Why spend more money for a Cisco or a Grandstream or even a Sipura?
Buy this now at $54.88 from our Ebay Store

Items are BRAND NEW and SEALED
Ready to Ship

Shipping Weight is 4 Pounds
For International Shipping, we use USPS
please click link for
international shipping cost.

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