8 Port Power Audio Amplifier gateway 8ports Asterisk Machine 8 sip gateway



Essential details
Model Number:
Power Amplification Gateway
Place of Origin:
Port Choice:
1/4/8/32 GOIP
Voip, VPN, IMEI change,SIM
SIP and H.323
2G Quad Band
Asterisk/ IP PBX
Voice Codec:
1 Year

Packaging & delivery

Selling Units:
Single item
Single package size:
20X18X5 cm
Single gross weight:
1.500 kg
Quantity(pieces) 1 – 5 6 – 10 >10
Lead time (days) 7 10 To be negotiated
Product Name: 8 Channels Power Amplification and Audio Amplification 

Model Name: GoIP-8

1. 8 Port channels,up to 8 cards
2. Quad band, IMEI changeable
3. Support of SMB32 SIM Bank, SIB128 SIM Band
4. VoIP SIP,H323,Remote Access
5. Optional SMI  termination
6. Easy to install and administrate
7. Auto Balance and Recharge(USSD)
8. Auto BTS changeable
9. Inter-calling, Inter-sending

Series Model: GoIP-1, GOIP-4, GOIP-8, GOIP-32, SINBANK-128

Our GoIPs come with all the common network, VoIP, and GSM features and they have been widely tested for compatibility, stability, and reliability. In addition, we’ve developed our proprietary Remote SIM technology for SIM card management without inserting SIM cards to GoIPs. Together with the SMS Server and SIM Sever, you can now build your own system for voice traffics between VoIP and GSM or a SMS Messaging system based on your application requirements.

In addition, each GoIP is equipped with a Remote Control client which enables a secure remote access method for technical support and GoIP management. All of the above features are available at no additional charges.
VoIP GSM Gateway GoIP-1 is a VoIP GSM Gateway for call termination (VoIP to GSM) and origination (GSM to VoIP). It is SIP&H.323 based and compatible with Asterisk, Trixbox, 3CX, SIP Proxy Server, VoipBuster. It can enable to make 1 call simultaneously from IP phones to GSM networks and GSM networks to IP phone.
         Key Features
Call from VoIP to GSM or GSM to VoIP
Bulk SMI
Auto check balance and recharge
Remote SIM Control
Outbound call from voip
Inbound call from gsm
Remote management

  Software Features

VoIP Protocols: SIP2.0
Audio Codecs: g711(alaw/ulaw), g729, g723.1
DTMF format: SIP INFO, RFC2833, Inband
Packet Loss Concealment
Programmable Jitter Buffer: Fixed, Dynamic, Adaptive
Network Connection: DHCP, PPPOE, Static IP
STUN Server
Support English, Chinese
Support Signaling encryption
Support media encryption
Support Media NAT Traversal
Support Remote control

Hardware Features

Channel: 8
Power Adapter: DC 12V/500MA
CPU: ARM9/300MHz
Flash: 4M
GSM Frequency: 850/900/1800/1900MHz
SIM Module (2G): M26: Support work in all countries which still support 2G Signal
Max power consumption: 5W
Weight: 1KG (Including AC/DC Adapter)
Size: 120mm (L) x 80mm (W) x 30mm (H)
Operating Temperature: 0 – 45℃
Working Humidity: 10% – 90% non condensation
Warranty: One year

   Free Software
SMS Server
SIM Server ( Sim Bank Scheduler Server )
Relay Server
Remote Access
The concept diagram below shows a simple VoIP network for call center application. It consists of a softswitch/IP PBX and VoIP clients such as an IP phone, a call center operator, and GoIPs. All VoIP clients are configured to register to the softswitch/IPPBX. For the GoIPs, each GSM channel is inserted with a valid SIM card in order to access the GSM network. Both incoming and outgoing calls between the VoIP and GSM/PSTN networks can now be realized


GSM Gateway 8ports can connect with Asterisk. Required features: the possibility to outgoing calls and receive incoming. For outgoing calls to have a choice of an arbitrary line.

Support work with 3CX system

In some places, because of the imperfect infrastructure, PSTN lines cannot be used. At this time, the gsm gateway can solve the problem of outbound calls of their telephone systems.

Imagine that you are on a long lasting business trip to the opposite side of the globe and you have to provide transparent and affordable voice connectivity with those on the other side where you came from. All modern means like Hangouts, Viber, Skype, Webex and others would suit only to technically advanced individuals or those who are used to the technologies of the 21st century. What about parents, grandparents, brothers, sisters, friends, or colleagues who don’t understand computers, don’t have
smartphones or don’t want to install special applications on their phones for the sake of a single phone call? gateway call will be help you make easy,


1: Call Origination And Termination for VoIP Traffic
1) Call Origination refers to a call initiated from the PSTN or cell phone network is terminated using device.
2) Call Termination refers to a call initiated as a call is terminated using PSTN or cell phone network.
3) As shown in the network topology diagram, a Service Provider is using s as call origination and termination devices.
– A call dialed to a gateway (right hand side) via GSM is first routed via and then terminated via a end point or
Service Provider.
– A call originated from the left hand side is routed to a gateway on the right hand side and then is dialed out as a GSM call.

Feedback from Customers
Company Introduction
CHINA SKYLINE TELECOM CO.,LIMITED was founded in 2002.We have over 200 employees ,20 departments and more than 10 production lines.More than 200 companies have established business with us,and they are from 40 different countries.

Our company is a manufacturer of SMS Gateway, SMS Modem, VoIP Gateways,GSM VoIP gateways, with well-equipped testing equipment and strong technical force. With a wide range, good quality, reasonable prices and stylish designs, our products are extensively used in communication industry and other industries. Our products are widely recognized and trusted by users and can meet continuously developing economic and social needs.

We welcome new and old customers from all walks of life to contact us for future business relationships and achieving mutual success!

Contact us!
Any inquiry, plz contact me freely.

Skype: live:alisa_1657
WhatsApp: +8613542062020