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GSM Voip Gateway With SIP And H323 Protocol Gsm Goip 32 Port Voip Gsm Gateway GoIP32 Voip Terminal

USD 1739.00USD 1850.00

GSM Voip Gateway With SIP And H323 Protocol Gsm Goip 32 Port Voip Gsm Gateway GoIP32 Voip Terminal

Description
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GSM Voip Gateway With SIP And H323 Protocol Gsm Goip 32 Port Voip Gsm Gateway GoIP32 Voip Terminal
Product Description
GSM GoIP32 VoIP Gateway
VoIP GSM Gateway GoIP bridges the GSM and the IP networks by enabling voice communications. It is installed on the Mobile SIM Card, you can register the GSM telephone on the VoIP Softswitch.it can be easily combined into arbitrary number of channels of Large Gateway Group. Low price, perfect sound quality and powerful function make GoIP GSM inevitable choice of system Integrators, traffic Business and soft switch Manufacturers.
Product Name: 32 Channels 32 SIMs GSM Gateway Model Name: GoIP-32 Specification: 1. 32 GSM channels,up to 32 sim cards 2. Quad band, IMEI changeable 3. Support of SMB32 SIM Bank, SIB128 SIM Band 4. VoIP SIP,H323,Remote Access 5. Optional SMS 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-16,GoIP-32, Simbank32, Simbank128 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.
Specification:
VoIP GSM Gateway GoIP-32 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 32 calls 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 SMS
 
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
GSM Channel: 32
 
Power Adapter: DC 12V/4.5A
 
CPU: ARM11/700MHz
 
RAM: 128MB
 
Flash: 8M
 
GSM Frequency: 850/900/1800/1900MHz
 
SIM Module (2G): Support work in all countries where still support 2G Signal
 
Max power consumption: 38W
 
Weight: 4KG (Including AC/DC Adapter)
 
Size: 520mm (L) x 280mm (W) x 200mm (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
Connections:
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.
Applications
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 VoIP. 2) Call Termination refers to a call initiated as a VoIP call is terminated using PSTN or cell phone network. 3) As shown in the network topology diagram, a VoIP Service Provider is using GoIPs as call origination and termination devices. - A call dialed to a GoIP (right hand side) via GSM is first routed via VoIP and then terminated via a VoIP end point or VoIP Service Provider. - A VoIP call originated from the left hand side is routed to a GoIP on the right hand side and then is dialed out as a GSM call.
 
2: IP PBX Call Origination and Termination
1) Instead of FXO gateways, GoIPs are as a call termination and origination device for the IP PBX as shown in the diagram above. 2) VoIP endpoints connected to the IP PBX can make calls to cellular/traditional telephone network via the GoIP GSM ports. 3) Outside callers can then call in via the GoIP GSM ports to reach any of the VoIP endpoints that are registered to the IP PBX. 4) GoIPs can be configured in a group mode such that all GSM ports can be used by just dialing only one GSM number. Please refer to the Call Center Application for more information.
 
3: Inbound Call Centre for Customer Service
1) Inbound call centre is operated by a company to administer incoming product support or information inquiries from consumers. 2) This application requires customers to call in via a single phone number. Each GSM channel in a GoIP has its own phone number. In order to achieve calling in via a single number, GoIPs must be configured in a phone group mode with one GoIP as a host and the others as clients. 3) An incoming call to the host GoIP is automatically forwarded to an idle GSM channel in the phone group. The call is then routed to a call attendant via a VoIP connection. 4) Key advantages: quick setup, scalable, portable, low system cost, low operating cost especially used in cellular network no charge on incoming.
 
4: Outbound Call Centre for Telemarketing/Solicitation
1) Outbound call centers are normally operated for telemarketing, solicitation of charitable or political donations, and debt collection etc. 2) In this application, only outgoing calls are made. GoIPs could be configured as SIP clients to registering to the SIP server/IPPBX or as a SIP Trunk to accept calls routed from the SIP Server/IP PBX. 3) Key advantages: quick setup, portable, scalable, low system cost, low operating cost.
 
5: Call Back Service
1. Call Back is referring to the telecommunications event that occurs when the originator of a call is immediately called back in a second call as a response. 2. GoIP could be used to achieve this function alone or as an terminal that is integrated in an existing call back server/platform. 3. For standalone operation,GoIP receives a call with caller ID information and then rejects the call immediately without answering the call.GoIP then calls back the caller so that he can dial a phone number to make a call.In this case,GoIP must register to a VoIP Service Provider who can offer terminate the call. 4. In a call back system,GoIP acts as a device to initiate the call back function.Typically, this is done in two ways. The first method is to send an SMS with the callee's phone number to the GoIP.The GoIP then sends both the caller's and callee's phone numbers to the call back server to complete the call back function.The second method is to call the GoIP and the hang up (with the call being answered).GoIP sends the caller's phone number to the call back server and the call back server calls the caller directly so that the caller can then dial a phone number to make a call.
 
6: Sending Bulk SMS Service
1. Sending bulk sms text messages is a common technique for telemarketing to reach the target customers. 2. A bulk SMS system can be implemented quickly and easily using GoIPs and our proprietary SMS server. Telemarketers are now have full control on how and when they want to send text messages. 3. In addition, SMS text messages are now used widely in many companies, organizations, schools, clubs as a mean for broadcasting information. They can now build their own SMS system without paying expensive charges to their GSM server provider. 4. This system can also take the advantage of using the same GSM service provider to send sms to the phone subscribers in the same service provider.
Gateway Web Management
How to log in Web Management?
Open the web browser and type the IP address. If it is the first time you login the gateway, please use the default settings below: IP Address: 192.168.8.1 Account: admin Password: admin
 
Following are the pages of GoIP web management:
Packing&Shipping
Packing List: (1) GoIP Gateway x 1 (2) Network Cable x 1
(3) Power Supply Adapter x 1 (4) Antennas x 32
Trade Terms
FOB, CFR, EXW, DAP
Payment Terms
Alibaba via Credit Card/Paypal, Bank Wire, T/T, Webmoney, Skrill, Western Union, Cash, USDT
Payment Condition
100% deposit in adavance
Delivery Time
1-3 days after full payment
Packing
Strong Carton Box with Water Proof
Shipping
DHL, UPS, EMS, FEDEX, ARAMEX, TNT or Other Way As You Require
FAQ
Q: 1 . What is main funtion of the GOIP 32 ports gateway? A: For call terminal and bulk sms.
And we have Live SMS Traffic to Vietnam, India, USA, Indonesia, Nigeria, Spain, Germany, France, Phillipines, Australia, New Zealand, Canada, UK, UAE, Bangladesh, Thailand, and European regions. If you have sim cards and interested in sms business for huge profit, please contact me!!
Q :2. How about our service? A: We have professional engineer to help you about all your tech problems anytime,the satisfaction of our customers is our greatest desire.
Q: 3.If my routes face the issue for quality not good , any solution? A: Firstly, you can check your internet, gsm signal, sim cards balance, sim blocking. Secondly, if everything is good, we will guide our engineer to help you check your configurations.
Q: 4.I need more ports , any device? A: If your ports are less than traffic, you should increase more ports to accpet more traffic. We have 4/8/16/32 ports and 4G SK gateway for you to expand your business.
Q: 5.How long we're warranty ? 1 year
Q: 6.During warranty time,if have problem, how to do? we'll test by teamview or send back, we'll exchange new in free.  
Q: 7.How long about our techinical support? in free,forever.
Q: 8. Quickly contact way? A: My skype ID: live:alisa_1657 Wechat/WhatsApp/Telegram: +86 13542062020
Related Products
 
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!
 
Specification

Brand Name : Vieruodis

Type : VoIP Gateway

Compatible Audio & Video Equipment Style : Earphones & Headphones,Speakers,Amplifier,Player,Digital-to-Analog Converter (DAC),Professional Audio Equipment,Stage Equipment,Recording Equipment,Display Equipment

Brand Name : SKYLINE

Certification : CE

Place of Origin : Guangdong, China

Model Number : GOIP32

Type : WIFI

Network : GSM

Origin : Mainland China

built-in : encryption

Voice Codec : G.723.1

color : White

Power supply : 12V/5A

Frequency : Quad Band

Network Protocols : HTTP

Compatible : Asterisk/ IP PBX

support : VPN, IMEI change,SMS

Warranty : 1 Year

protocol : SIP and H.323

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