equipment for call terminal

Dear voip partner :
We have voip GSM/CDMA gateaway for call terminal .
Key Features
--Multiple GoIP grouping mode
--Provide 1/4/8/16 cellular channels for IP-PBX
--Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
--Single or Multiple Server Registrations
--Two 10/100 Ethernet circuits connect to the LAN and an additional device
--GSM module for making GSM calls
--Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
--VLAN and QoS support  voip sim box
--NAT Transversal and Router functions
--Voice prompts, HTTP Web, Auto Provision support for configuration and updates
--Highly stable embedded Linux operating system in high performance ARM 9 Processor

Basic Features
--LEDs for Power, Ready, Status, WAN, PC, GSM
--Call forward from GSM to VoIP and VoIP to GSM
--Dial in mode or dial out mode only
--Dial Plan 16 port gsm voip gateway
--Password protection for both GSM dial in or dial out
--Retransmit GSM Caller ID to VoIP terminal

Enhanced Features
--Dynamic selection of codec
--Advanced jitter buffer
--Automatic traversal of NAT and firewall
--VLAN / Qos
--Router 16 port gsm voip gateway
--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

Hardware Specifications
--Processor: ARM9E 133MHz
--DSP: VPDSP101 196MHz
--Memory: RAM 16MB/ Flash 4MB
--GSM Module: Type: 850MHz, 900MHz, 1800MHz, 1900MHz
--Power: Input AC100V ~ 240V, output DC12V/2A +-10%
--Power consumption: 32W maximum
--Network card: 100/10Base-T x2
--LED: Operation and lines light
--GSM Passway:16
--Operating temperature: 10��C to 40��C (32��F to 104��F)
--Storage temperature: 0��C to 50��C (32��F to 122��F)
--Working Humidity: 40% ~ 90% Not congealed
--Weight: 1203 g (1 lb) (Including AC/DC Adapter)
--Warranty: one year

Supported Standards
--ITU: H.323 V4, 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



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