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Jan-19-2026 0.0 руб. 0.39 руб. 0 руб.
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support 4 sim cards port gsm voip gateway/pbx sip gateway GoIP works with IP PBX/Asterisk / goip4 | Мобильные телефоны иsupport 4 sim cards port gsm voip gateway/pbx sip gateway GoIP works with IP PBX/Asterisk / goip4 | Мобильные телефоны и


4 channels GSM Wireless VoIP Gateway GoIP Gateway

GoIP4 ALL.jpg

The 4 channels GSM VoIP gateway is a 4 SIM Card Broadband Phone Gateway that had been developed by Hi Telecom LTD.  GOIP_4  SIM Card Broadband Phone Gateway is a new product that connect the GSM and the VOIPseamlessly. To GOIP_4 what is installed on the Mobile SIM Card, you can register the GSM telephone on the VOIP Softswitch.SIP and H.323 agreement are built in the GOIP_4 and configured flexible. Caller ID can be seen by using SIP. Flexible routing can meet the need of all kinds of call forwarding; even more special is that GOIP_4 support multi-device group, it can be easily combined into arbitrary number of channels of Large Gateway Group.

 

Key Features

  • Multiple GoIP4 grouping mode
  • Provide four 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
  • 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 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: 12W maximum
  • Network card: 100/10Base-T x2
  • LED: Operation and lines light
  • GSM Passway:four
  • 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: 450 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

 

Free roaming:

In order to promote our product ,Now we have added some new funcitions in  GoIP.By adding the new funcitions,you can build the call without using the softwhich or platform,just depending on the internet.it makes our calls more convenient and easier.what’s more, it can save a lot of call fee.Here is the following example:

 

Example: 
peer to peer:it is a new function that you can use our voip without platform,what is more,it is free roaming for international call.you just pay the local call

Model 1:

 1.jpg

Model 2:

 2.jpg

 

Model 3:

 3.jpg

Model 4:

 4.jpg

Common function:

1 PSTN to VoIP

 5.jpg

Description:using the GoIP to connect with VoIP

2 VoIP to PSTN

 6.jpg

Description: using the GoIP to connect with PSTN

3 Calling forward

 7.jpg

Description:If you are in china but your main business is in Malaysia, you only put a Malaysia SIM card into the GoIP.In this condition,all calling the SIM number can connect your Phone number in China directly.

4 calling back

 8.jpg

Description:When you are using the telephone, and you are want to get  preferential from VoIP Phone anytime and anywhere.You just call the SIM card number which in GoIP,the calling number would send to Server by GoIP,then the server receive the calling number and establish the new calling.Then you will receive the new calling,you just accept the calling,Now you are calling with your customer by server

GoIP4-1.jpgGoIP4-2.jpgGoIP4-5.jpg

   


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