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GSM VoIP Gateway
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  • Overview
  • Specification
  • Inquire
  • Four-port GSM Gateway with internal antenna (PEGI004) provides the extension of GoIP and supports up to 4 GSM channels.

    Each GSM channel can be programmed individually to establish the connectivity with IP-PBX, or can be bundled as a single trunk in connection with IP-PBX.

    It is ideal for VoIP to wireless services where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. It brings mobility, scalability and significant savings on long distance charges.


    Key features
    1. Provide four cellular channels for IP-PBX
    2. Support open standard SIP Protocols (IETF SIP V2)
    3. Support SIP proxy mode
    4. Multiple GoIP4 grouping mode
    5. With internal antenna
    6. Two 10/100 Ethernet ports for the LAN and an additional device
    7. Guad band GSM module: support GSM 850MHz, 900 MHz, 1800 MHz, 1900MHz
    8. Speech quality ensured by QoS at LAN, IP layers and comprehensive jitter buffer
    9. VLAN and QoS support
    10. NAT Transversal
    11. Voice prompts, HTTP Web
    12. Highly stable embedded Linux operating system in high performance ARM 9 Processor 

  • Hardware
    1. Processor: ARM9E 133MHz
    2. DSP: VPDSP101-4 100MHz
    3. Memory: RAM 16MB/ Flash 4MB
    4. GSM Module: Type: 850MHz, 900MHz, 1800MHz, 1900MHz
    5. Power: Input AC100V ~ 240V, output 12V dc 2000 mA
    6. Power consumption: 5W maximum
    7. Operating temperature: 10°C to 40°C (32°F to 104°F)
    8. Storage temperature: 0°C to 50°C (32°F to 122°F)
    9. Working Humidity: 40% ~ 90% not congealed
    10. Weight: 0.9kg (3 lb) (Including AC/DC Adapter)
    11. Size: 195mm (W) x 340mm (L) x 57mm (H)
    12. Power: 12 V 2A.

    Enhanced Features
    1. LEDs for Power, Ready, Status, WAN, PC, GSM
    2. Call forward from GSM to VoIP and VoIP to GSM
    3. Dial in mode or dial out mode only
    4. Dial Plan
    5. Password protection for both GSM dial in or dial out
    6. Retransmit GSM Caller ID to VoIP terminal
    7. Dynamic selection of codecs
    8. Advanced jitter buffer
    9. Automatic traversal of NAT and firewall
    10. VLAN / Qos
    11. Echo cancellation for Speakerphone
    12. Comfort noise generation (CNG)
    13. Voice activity detection (VAD)
    14. Auto provisioning (requires auto provisioning server)
    15. Firmware upgrade from GUI


    1. ITU: H.323 V4, H.225, H.235, H.245, H.450
    2. RFC 1889 - RTP/RTCP
    3. RFC 2327 - SDP
    4. RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
    5. RFC 2976 - SIP INFO Method
    6. RFC 3261 - SIP
    7. RFC 3264 - Offer/Answer model with SDP
    8. RFC 3515 - SIP REFER Method
    9. RFC 3842 - A Message Summary and Message Waiting Indicator
    10. RFC 3489 (STUN) - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
    11. RFC 3892 - SIP Referred-By Mechanism
    12. Proprietary Firewall-Pass-Through Technology
    13. Codec: G.711 (A/µ law), G.729A/B, G.723.1
    14. DTMF: RFC 2833, In-band DTMF, SIP INFO
    15. Web-based Management
    16. PPP over Ethernet (PPPoE)
    17. PPP Authentication Protocol (PAP)
    18. Internet Control Message Protocol (ICMP)
    19. TFTP Client
    20. Hyper Text Transfer Protocol (HTTP)
    21. Dynamic Host Configuration Protocol (DHCP)
    22. Domain Name System (DNS)
    23. User account authentication using MD5

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Unit 1311, 13/F, Building C, Jufu Garden, 41 Guowei Road, Luohu District, Shenzhen China 518004