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