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GOIP-1-S 1 1 port gsm and 1 channel FX gateway for call terminal

 
GOIP-1-S 1 1 port gsm and 1 channel FX gateway for call terminal
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GOIP-1-S 1 1 port gsm and 1 channel FX gateway for call terminal

OUR PRICE : 163.20  Incl. VAT
 136.00  Excl. VAT
 
 
Discount Available for orders of 10 units or more - call +44 (0) 207 096 8420
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GOIP-1-S 1 1 port gsm and 1 channel FX gateway for call terminal

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DESCRIPTION  
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GOIP-1-S 1 1 port gsm and 1 channel FX gateway for call terminal

One GSM Port with one FXS port VoIP Gateway GS-1 bridges the GSM,Analog telephone,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/ law), G.729A/B, G.723.1
  • DTMF: RFC 2833, In-band DTMF, SIP INFO

Physical and Environmental

Operating temperature: 10C to 40C (50F to 104F)

Storage temperature: 0C to 50C (32F to 122F)

Weight: 0.64 KG (1.5 lb) (Including AC/DC Adapter)

Size:105 mm (W) x 100 mm (L) x 28 mm (H) (Width x Length x Heigth

Power: 12 Vdc 500 mA (AC/DC adapter included)

Safety:

Emissions:

Immunity:

Other Approval:

Warranty: one year


Take2teq Ltd, Charnwood House, Longdown Road, Farnham, Surrey,
GU10 3JL, United Kingdom

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