Business Type:
Trading Company
Business Range:
4G LTE Gateway, GOIP GSM VoIP Gayeway, SIM BANK, RoIP Radio Gateway, FXS VoIP Gateway
Establishment:
2005
R&D Capacity:
OEM, ODM, Others
Terms of Payment:
LC, T/T, D/P, Paypal, Western Union
Main Markets:
Mid East, Eastern Europe, Southeast Asia
OEM/ODM Service
Sample Available

Shenzhen Hybertone Technology Co., Ltd. is a new and high-tech enterprise, which integrates scientific research, development, production and sale. Since we were established, we have been engaged in th...

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General Supplier

Gsm Voip Goip Gateway Sip Trunk To Asterisk Ip Pbx Goip8

Get Latest Price
Min. Order / Reference FOB Price
1 Piece US $226.00/ Piece
Local Area: Shenzhen, Guangdong, China
R&D Capacity: OEM, ODM, Other
Payment Terms: LC, T/T, D/P, Paypal, Western Union
Brand: Hybertone
Model Number: GOIP8
Type: VoIP Gateway
SIM CARD PORT: 8
Key Features
Provide 8 cellular channels for IP-PBX
Open Standard VoIP Protocols (SIP&H.323)
Single or Multiple Server Registrations
Two 10/100 Ethernet for WAN / LAN connections
Peer-to-Peer IP Calls
Quad band GSM module: 850MHz, 900 MHz, 1800 MHz, 1900MHz
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
Line Echo Cancellation
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 ProcessorEnhanced Features
LEDs for Power, Ready, Status, WAN, PC, GSM
Dial in mode or dial out mode only
Call forward from GSM to VoIP and VoIP to GSM
Dial Plan
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal
Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
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 ChineseSupported 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 (STUN)- Simple Traversal of 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
Web-base Management
PPP over Ethernet (PPPoE)
PPP Authentication Protocol (PAP)
Internet Control Message Protocol (ICMP)
TFTP Client
Hyper Text Transfer Protocol (HTTP)
Dynamic Host Configuration Protocol (DHCP)
User account authentication using MD5Free Software
SMS Server
SIM Server ( Sim Bank Scheduler Server )
Relay Server
Remote Access
Hardware Specifications
Processor: ARM9E 133MHz
DSP: VPDSP101-4 100MHz
Memory: RAM 16MB/ Flash 4MB
GSM Module: 850MHz, 900MHz, 1800MHz, 1900MHz
Power: 12 VDC 3A (110V-220V) (AC/DC adapter included)
Power consumption: 20W maximum
Operating temperature: 10°C to 40°C (32°F to 104°F)
Storage temperature: 0°C to 50°C (32°F to 122°F)
Size: 200mm (W) x 370mm (L) x 72mm (H)
Weight: 1.2KG (Including AC/DC Adapter)
Warranty: one yearNotes:
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.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.

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