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GoIP-4 4 SIM VoIP GSM Gateway IMEI change Bulk SMS For Asterisk PBX SIP Server,GoIP ATA GoIP SIP Gateway,GSM Phone System Appliance

  • Item : Voip GSM gateway
  • Model No.: GOIP-4
  • Bands: GSM 850/900/1800/1900mhz
  • Supports H.323 V4 & SIP V2
  • Supports Sim Bank
  • Supports IMEI change
Overview
Specs
Reviews
Item : Voip GSM gateway
Model No.: GOIP-4
Bands: GSM 850/900/1800/1900mhz
Supports H.323 V4 & SIP V2
Supports Sim Bank
Supports IMEI change

The GoIP-4 is 4 Channels GSM Gateway bridges the GSM and the IP networks by enabling voice communications. It is an ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via a VoIP network. It enable you can save distance charges significantly .

Key Features:
-GOIP-4 support Open Standard VoIP Protocols H.323 V4 and SIP V2
-It has 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
-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
-It support VLAN and QoS
-Single or Multiple Server Registrations


Basic Features:


-LEDs for Power, Ready, Status, WAN, PC, GSM
-Support 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


- 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

Physical and Environmental:


-Operating temperature: 10°C to 40°C (50°F to 104°F)
-Storage temperature: 0°C to 50°C (32°F to 122°F)
-Weight:0.5 kg (Including AC/DC Adapter)
-Size:160 mm (W) x 252 mm (L) x 30 mm (H)
-Power: 12 Vdc 3A (AC/DC adapter included)

Warranty & Returns

Warranty, Returns, And Additional Information

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