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Shenzhen DBL Technology Limited

Gsm Voip Gateway, Voip Gateway, Goip manufacturer / supplier in China, offering 1-GSM&1-FXS Port VoIP Gateway GS-1, GoIP 8 8 Channel VoIP GSM Gateway/ 8 SIM Card Wireless VoIP Terminal Support SMS Server, 8 Port/Channel GSM VoIP Gateway with SIP&H. 323 (GoIP8) and so on.

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Supplier Homepage Product VoIP GSM Gateway 1-GSM&1-FXS Port VoIP Gateway GS-1

1-GSM&1-FXS Port VoIP Gateway GS-1

FOB Price: Get Latest Price
Min. Order: 1 Unit
Production Capacity: 10, 000 PCS/Month
Transport Package: Box Dimension: 22*17.5*6cm (L*W*H) G.W.:0.40kg
Payment Terms: T/T, Western Union, Paypal
Basic Info
  • Model NO.: GS-1
  • Port: 1 GSM Port and 1 FXS Port
  • VoIP Protocal: H. 323&SIP
  • Working Frequency: Default 900m/1800m /Optional 850m/1900m
  • Warranty: One Year
  • Specification: Package: 20PCS/Carton
  • Type: Gateway
  • Support: Vlan and Qos, DHCP, Pppoe, Fix IP
  • Build in: Reply Encryption
  • IMEI Change: Support
  • Trademark: DBL
  • Origin: Guangdong China (Mainland)
Product Description
The 1-GSM&1-FXS Port VoIP Gateway GS-1 is a new product developed by DBL.

The DBL High Performance VoIP GSM Gateway GoIP (1-SIM Card Port) bridges the GSM and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized.

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

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
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