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  • Modular Design IP PBX for SMB
  • Remote office Centralized Management solution
  • 3rd party app integration, Enterprise Billing, Android & iOS client
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  • Cost-effective IP-PBX Solution for SMB
  • FXS, FXO, GSM, BRI and PRI VoIP Gateways
  • Rich features and reliable performance
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goip gateway

GSM VoIP Gateway ( goip gateway ) for call termination, HyberTone Technology

business card.jpg
Models of GSM VoIP Gateway ( goip gateway )
Series 1 SIM port 4 SIM ports 8 SIM ports 16 SIM ports32 SIM ports
External AntennaGoIP-1 GoIP-4 GoIP-8 GoIP-16 GoIP-32
Internal AntennaGoIP-1i GoIP-4i



1, 4, 8, 16, 32 ports GSM VoIP Gateway ( goip gateway ) ( External Antenna series )

GoIP-1_300.jpg
GoIP-4_300.jpg
GoIP-8_300.jpg
GoIP-16_300.jpg

GoIP-32-5.jpg


1, 4 ports GSM VoIP Gateway ( goip gateway ) ( Internal Antenna series )

GoIP-1i_300.jpg
GoIP-4i_300.jpg



Specification


Basic Features:

    • For call termination (VoIP to GSM) and origination (GSM to VoIP) }
    • Standard SIP & H.323 protocol
    • GSM: quad-band 850/900/1800/1900MHz

Major Advantages:

    • Lightweight and Portable
    • IMEI Changeable
    • GSM Base Station Optional
    • Support SIM Bank/ SIM Sever
    • Manual/ Automatic Selection Operators
    • Sending and Receiving SMS and USSD (Web Interface)

Free Software/ Sever:

    • Remote Control Server: Access Interface Remotely

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

    • SMS Server:
      • Send Bulk SMS
      • Provide CDR and ASR
      • Auto Balance and Recharge

    • SIM Server:
      • Rotate SIM Cards on Duty
      • Set GSM Group (Assign several SIMs Per GSM Port)
      • Set Talk Time per SIM, Set Day of week, Set Time Range
      • Monitor CDR, ASR, ACD

Key Features


    • Provide 1, 4, 8, 16 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 Processor


Enhanced 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 Chinese


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 (STUN)- 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
    • 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 MD5


Lead Sales: Leo

Website: www.hybertone.com

Email: Leo@hybertone.com

Skype: Leohybertone

Tel: +86 18576773508


Facebook
http://www.facebook.com/HyberTone.VoIP

Youtube
http://www.youtube.com/HyberTone


goip gateway

GSM VoIP Gateway ( goip gateway ) for call termination, HyberTone Technology

business card.jpg
Models of GSM VoIP Gateway ( goip gateway )
Series 1 SIM port 4 SIM ports 8 SIM ports 16 SIM ports32 SIM ports
External AntennaGoIP-1 GoIP-4 GoIP-8 GoIP-16 GoIP-32
Internal AntennaGoIP-1i GoIP-4i



1, 4, 8, 16, 32 ports GSM VoIP Gateway ( goip gateway ) ( External Antenna series )

GoIP-1_300.jpg
GoIP-4_300.jpg
GoIP-8_300.jpg
GoIP-16_300.jpg

GoIP-32-5.jpg


1, 4 ports GSM VoIP Gateway ( goip gateway ) ( Internal Antenna series )

GoIP-1i_300.jpg
GoIP-4i_300.jpg



Specification


Basic Features:

    • For call termination (VoIP to GSM) and origination (GSM to VoIP) }
    • Standard SIP & H.323 protocol
    • GSM: quad-band 850/900/1800/1900MHz

Major Advantages:

    • Lightweight and Portable
    • IMEI Changeable
    • GSM Base Station Optional
    • Support SIM Bank/ SIM Sever
    • Manual/ Automatic Selection Operators
    • Sending and Receiving SMS and USSD (Web Interface)

Free Software/ Sever:

    • Remote Control Server: Access Interface Remotely

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

    • SMS Server:
      • Send Bulk SMS
      • Provide CDR and ASR
      • Auto Balance and Recharge

    • SIM Server:
      • Rotate SIM Cards on Duty
      • Set GSM Group (Assign several SIMs Per GSM Port)
      • Set Talk Time per SIM, Set Day of week, Set Time Range
      • Monitor CDR, ASR, ACD

Key Features


    • Provide 1, 4, 8, 16 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 Processor


Enhanced 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 Chinese


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 (STUN)- 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
    • 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 MD5


Lead Sales: Leo

Website: www.hybertone.com

Email: Leo@hybertone.com

Skype: Leohybertone

Tel: +86 18576773508


Facebook
http://www.facebook.com/HyberTone.VoIP

Youtube
http://www.youtube.com/HyberTone


Created by: hybertone, Last modification: Tue 16 of Sep, 2014 (02:19 UTC) by audi
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