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Home » Product List » GSM Fixed Wireless Terminal » DBL GOIP 8 Ports GSM VoIP Gateway/Wireless Terminal/Wll Termination

Product Details

DBL GOIP 8 Ports GSM VoIP Gateway/Wireless Terminal/Wll Termination

   
Packing:  20PCS/CTN
Model NO.:  GOIP 800
Standard:  252*160*30mm
Productivity:  10, 000PCS/Month
Trademark:  Etross
Origin:  China
Usage:  Laptop
Type:  VoIP Products
Style:  Wireless
Working Frequency:  850/900/1800/1900MHz
Back-up Battery:  Available
Port Number:  2 Ports
CE:  Available
Export Markets:  North America, South America, Eastern Europe, Southeast Asia, Africa, Oceania, Mid East, Eastern Asia, Western Europe
Product Description

8 VoIP & 8 GSM Channels

The GoIP800 GSM Gateway 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 a VoIP network. Significant savings on long distance charges can be realized.

Specifications
1. Dynamic selection of codec
2. Advanced jitter buffer
3. VLAN / Qos
4. Router

Brand Name: Etross or OEM Brand

Product Name: GSM VOIP GATEWAY 800

Key Features:

1. Open Standard VOIP Products(ITU H. 323 V4 and IETF SIP V2)

2. Single or Multiple Server Registrations

3. Two 10/100 Ethernet circuits connect to the LAN and an additional deivice

4. GSM module for making GSM calls

5. Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer

6. VLAN and QoS support

7. NAT Transversal and Router functions

8. Voice prompts, HTTP Web, Auto Provision support for configuration and updates

8. Highly stable embedded Linux operating system in high performance ARM 9 Processor

Basic Features:

1. LEDs for Power, Ready, Status, WAN, PC, GSM

2. Call forward from GSM to VOIP and VOIP to GSM

3. Dail in mode or dail out mode only

4. Dail Plan

5. Password protection for both GSM dial in or dial out

6. Retransmit GSM Caller ID to VoIP terminal

Enhanced Features:

1. Dyanamic selection of codec

2. Advanced jitter buffer

3. Automatic traversal if NAT and firewall

4. VLAN/Qos

5. Echo cancellation for Speaker phone

6. Comfort noise generation(CNG)

7. Voice activity detection(VAD)

8. Auto provisioning ( requires aito provisioning server)

9. On line firmware upgrade

10. Multi0-language support English and Chinese


Supported Standards

1. RFC 2976 - SIP INFO Method

2. RFC 3261 - SIP

3. RFC 3264 - Offer/Answer model with SDP

4. RFC 3515 - SIP REFER Method

5. RFC 3842 - A Message Summary and Message Waiting Indicator

6. RFC 3489 - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)

7. RFC 3891 - SIP 'Replaces' Header

8. RFC 3892 - SIP Referred-By Mechanism

9. Draft-ietf-sipping-cc-transfer-04 - Session Initiation Protocol Call Control - Transfer

10. Codec: G. 711 (A/µ Law), G. 729A/B, G. 723.1

Physical and Environmental:

1. Operating temperature: 10 to 40 degree
2. Storage temperature: 0 to 50 degree
3. Power: 12 Vdc 500 mA (AC/DC adapter included)
4. Warranty: One year

 
 

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