GOIP1, 1 channel voip gsm gateway/ goip voip gateway call termination for PBX, Asterisk

Price:

$155

Quantity Order:

Origin:

China

Pack. & Delivery:

10 pieces / carton 35.7*28.2*35.5cm

Overview

GoIP GSM Gateway bridges the GSM and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line ( PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized.

Key Features
--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
--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
--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
--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.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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