VoIP GSM FXS Gateway PSTN

VoIP GSM FXS Gateway PSTN

Negotiable /Piece

Min.Order:1 Piece

Supply Ability:
0 Piece / Pieces per Month
Payment Terms:
T/T L/C D/P D/A
Delivery Detail:
7 days

Quick Details View All >

Brand Name:
DBL
Place of Origin:
China
Model Number:
GS-1

DBL Technology Co., Limited

Business Type: Trading Company
Main Products: VoIP GSM Gateway ,GoIP GSM Gateway ,Asterisk GSM Gateway

Product Details

Delivery Time:3-5 workdays

VoIP GSM FXS Gateway PSTN
 
 
Quick detail:
 
Place of origin: Guangdong, China
Brand name: DBL
Model Number: GS-1
Type: VoIP  Gateway
Ports: 1-GSM&1-FXS
Size: 13*10*6cm
Protocol: SIP & H.323
Support: voip, gsm
 
 
Description:
 
GS-1 is a GoIP-1 with one FXS port VoIP Gateway , it bridges the GSM,Analog telephone,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. Low price, perfect sound quality and powerful function make GoIP GSM inevitable choice of system Integrators, traffic Business and soft switch Manufacturers.
 
Application
 
From VoIP to PSTN

 

2, PSTNàGoIPàPSTN( without long distance charge.)
 

 
Competitive Advantage:
 
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
 
 
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: 0.42 KG (Including AC/DC Adapter)
Size: 21.4*16.8*5.5cm
Power: Input 100-240V~50/60HZ 0.4A Max   
             Output:12V-1A
Warranty: one year

Contact Supplier

Mr. Banner sale Manager Chat Now
Telephone
86-755-88290233
Mobile
86-13266594560
Fax
86-755-88291220
Skype
sasadbl
Address
Room 5C,Aozhihao Integrated Building, Xinzhou 4th Street,Futian District Wenzhou,Zhejiang

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