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4 lines VOIP SIP Phone VLAN Support, VoIP IP Phone for Office EP-8201

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Product ID: 512851102

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Item specifics

Place of Origin:: China (Mainland)

Wire Type:: Corded

Brand Name:: Yagle

Model Number:: EP-8201

Type:: VoIP Phone

DTMF::: RFC 2833, In-band DTMF, SIP INFO

Message waiting indicator::: Router

support::: VLAN and QoS

Product Description

4 Port FXO VoIP Gateway

Overview

The EP-8201 is a high quality phone with lots of features for both business and residential users. Its slim and upright design makes it an ideal desktop phone. The phone is based on ITU-H.323 V4 and IETF SIP V2 open standards. The two protocols approach makes the phone to be compatible to most VoIP systems in deployment today. The Phone is designed for the ease of installation and setup. The PoE option simplify the installation in a PoE LAN environment. In addition, the second Ethernet Port allows the existing PC to be connected to the phone directly without addition an additional Ethernet Hub or Switch. Various configuration modes allow the user / system administrator to configure the phone automatically or quickly.

	For both business and residential users
	Has a graphical LCD with back light, programmable keys
	2 Ethernet Ports (with optional PoE),4-call appearances
	Headset port
	The phone is already CE, FCC, and CCC approved
	supports Standard SIP V2
Highlights	Built-in SIP V2 Protocol
	Built-in 3-way conferencing
	4 independent lines
	Support up to 4 different servers
	Auto provisioning and firmware updates
	Encryption transversal
	Multiple languages
	Large 128*64 LCD display
Key Features	Open Standard VoIP Protocols (IETF SIP V2)
	All standard PBX functions
	Four call appearances support two simultaneous calls
	Two 10/100 Ethernet circuits connect to the LAN and an additional device
	Graphical LCD
	Full featured and programmable keypad for all phone functions
	Phone display in English and Chinese (Other languages available upon request)
	Buttons and keys for all commonly used functions
	Message waiting LED
	Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
	Full duplex speaker phone
	VLAN and QoS support
	NAT Transversal and Router functions
	Power over Ethernet (PoE) or AC/DC adapter
	Menu, HTTP Web, Auto Provision support for configuration and updates
	Highly stable embedded Linux operating system in high performance ARM 9 Processor
Basic Feature	Call forward
	Call transfer
	Call hold
	Mute
	Redial
	Display caller ID
	Display call duration
	Display date and time
	SMS Capable
	Access voice mail
	Send DTMF tones
	Message waiting indication (MWI)
	100 phone book entries
	30 most recent call records for dialled, incoming, and missed calls
	Adjustment of LCD contrast (4 levels)
	Adjustment of handset volume (6 levels)
	Adjustment of speaker phone volume (6 levels)
Enhanced Features	Dynamic selection of codec
	Advanced jitter buffer
	Automatic traversal of NAT and firewall
	VLAN / Qos
	Router
	Echo cancellation for Speakerphone
	Comfort noise generation (CNG)
	Voice activity detection (VAD)
	Auto provisioning (requires auto provisioning server)
	On line firmware upgrade
	Multi-language support: English and Chinese
Supported Standards	ITU: H.225, H.235, H.245, H.450
	RFC 1889 - RTP/RTCP
	RFC 2327 – SDP
	RFC 2833 – RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
	RFC 2976 – SIP INFO Method
	RFC 3261 – SIP
	RFC 3264 – Offer/Answer model with SDP
	RFC 3515 – SIP REFER Method
	RFC 3842 – A Message Summary and Message Waiting Indicator
	RFC 3489 – Simple Traversal of User Datagram Protocol (UDP)
	Through Network Address Translators (NATs)
	RFC 3891 – SIP “Replaces” Header
	RFC 3892 – SIP Referred-By Mechanism
	draft-ietf-sipping-cc-transfer-04 – Session Initiation Protocol Call Control - Transfer
	Codec: G.711 (A/μ law), G.729A/B, G.723.1
	DTMF: RFC 2833, In-band DTMF, SIP INFO