GSM to VoIP (SIP) Cellular Gateway
1 x GSM Cellular Channel
Support quad bands – 850/900/1800/1900 MHz
Support STUN/NAT Traversal
SIP Standard Compatible
Support Peer-to-Peer mode
Support SIP proxy mode
Support Line-Hunt
Password or trust list protection for dial-in and dial-out
Support Text messaging
Compatible with Panasonic IP-PBX and Asterisk (including Trixbox, Fonality, Switchvox, FreePBX) - We have tested with Panasonic IP-PBX and Asterisk based IP-PBX and it's fully compatible.
GoIP GSM Gateway bridges the GSM services and the IP networks. It is ideal for VoIP to wireless services where a fixed telephone line (PSTN) is not available or for cell phone roaming via the a VoIP network.
GoIP GSM Gateway is designed to work in conjunction with key phone systems and IP-PBX to provide GSM communications. The extensive compatibility of the GoIP makes it an ideal choice to be deployed in multi-vendor open architecture VoIP networks.
GoIP GSM Gateway is a great way to provide fast phone service deployment where regular PSTN line may not be readily available. GoIP GSM gateway also provides significant savings in usage (long distance or international), infrastructure and maintenance cost compared to conventional PSTN.
Benefits of GoIP
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Extensive product compatibility with industry leading vendors
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Cost-Savings on phone calls between mobiles or to PSTN
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Easy to install – IP device with Web based management interface
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Can be managed and monitored remotely over Internet - a great service to offer to customers by system integrators
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GoIPs can be grouped together to establish GSM gateway cluster
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Termination between GSM/VoIP
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Schedule or on-demand SMS Broadcast messages to users (additional SMS server is required)
Key Features
Key Features
- Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
- Single or Multiple Server Registrations
- Two 10/100 Ethernet circuits connect to the LAN and an additional device
- GSM module for making GSM calls
- Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
- 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
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Basic Features
- LEDs for Power, Ready, Status, WAN, PC, GSM
- Call forward from GSM to VoIP and VoIP to GSM
- Dial in mode or dial out mode only
- Dial Plan
- Password protection for both GSM dial in or dial out
- Retransmit GSM Caller ID to VoIP terminal
Enhanced Features
- Dynamic selection of codec
- Advanced jitter buffer
- Automatic traversal of NAT and firewall
- VLAN / Qos
- Router
- 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 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
Physical and Environmental
Operating temperature: 10°C to 40°C (50°F to 104°F)
Storage temperature: 0°C to 50°C (32°F to 122°F)
Weight:500 g (1 lb) (Including AC/DC Adapter)
Size:105 mm (W) x 100 mm (L) x 28 mm (H) (Width x Length x Heigth
Power: 12 Vdc 500 mA (AC/DC adapter included)
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