Low cost 8 port goip gsm gateway goip gsm 8 sip GSM GOIP8 gsm gateway 8 sims goip gateway voip 8

Low cost 8 port goip gsm gateway goip gsm 8 sip GSM GOIP8 gsm gateway 8 sims goip gateway voip 8
    GoIP 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 8 calls simultaneously from IP phones to GSM networks and GSM networks to IP phone.Specification:

1. 8 GSM channels,up to 8 sim cards
2. Quad band ,IMEI changeable
3. Support of SMB32 SIM Bank
4. VoIP SIP,H323,Remote Access
5. Optional SMS termination
6. Easy to install and administrate
7. Auto Balance and Recharge
8. Auto BTS changeable

Series Model: GoIP-1,GoIP-4,GoIP-8,GoIP-16,GoIP-32

8 ports and 8 sims goip gateway

Main Functions

· GSM Carriers and Base Stations Optional

· Sending SMS / USSD via built-in webpage or SMS Server

· Remote SIM

· Auto balance check and recharge (prepaid SIM)

· Call time restrictions

· Remote Control Management

· Relay Server support for NAT and encryption of IP packets

· PPTP VPN and Relay Encryption

· Built-in SIP Proxy for peer-to-peer operation

· GSM Quad Band: 850 / 900/1800/1900 MHz

· Call forward from GSM to VOIP and VOIP to GSM

· Call Back

Main Advantages:

· Lightweight and Portable

· Easy to install and administrate

· IMEI Changeable

· GSM Base Station Optional

· Support SIM Bank/ SIM Sever

· Manual/ Automatic Selection Operators

· Sending and Receiving SMS and USSD (Web Interface)

· Flexible Dial plan fo number alternation, call screen.

· Flexible configuration modes.

· All functions can be set on web.

· Provide CDR.

Free Software/Server

Remote Control Server----Access Interface Remotely.

Relay Server----Relay Encryption( Make Terminals Traversal the NAT without STUN and Outbound Proxy)

SMS Server----Remote SIM cards; Send Bulk SMS; Provide CDR and ASR; Auto Balance and Recharge.

SIM Server----Remote SIM Cards on Duty; Set GSM Group( Assign several SIMs per GSM Port); Set talk time per SIM, Set Day of week, time range; Monitor CDR, ASR, ACD.

Autocfg Server---- Maintence all DBL termial device anywhere; embeded remote control server; Remote sim auto configuration.

Protocols: SIP/H.323

 TCP/IP:IP/TCP/UDP/RTP/RTCP/,CMP/ARP/RARP/SNTP, DHCP/DNSClient,IEEE802.1P/Q.ToS/DiffServ. NAT Traversal, STUN, UPNP.
IP Assignment, Static IP, DHCP, PPPoE

Picture show 

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/IP PBX.  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.

wireless voip phone

Solutions

1: Call Origination And Termination for VoIP Traffic
Notes: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
Notes: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
Notes: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 calls.

4: Outbound Call Centre for Telemarketing/Solicitation
Notes:
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/IP PBX 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
Notes: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
Notes: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.

 Packaging & Shipping

Company Information

Price and support

 we are real manufacturer and can give you factory price. we can give you professional free support services with the help of our technology support engineers.                                  

 Our Services

 >We provide stronge tech support about VOIP server and A-Z termination. (7 days*24 hours)

>We can help you rent a VoIP server in USA.

>We can provide copy version VOS.

>We can install Relay server, SMS server,SMB server for you. (for free)

>We can offer voip terminal guide.

>We may need your termina routes.

>We offer full tech support untill your gateway runs.

For more details plz contact:

Skype: Robert*@chinaskyline.net, (Remove *)

WhatsA/wechat: +86 13534193370

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