Home > Products > GSM VOIP GATEWAY/GOIP > GSM VoIP Gateway 4 Port/GoIP Gateway 4 Ports for Call Termination to Save Call Cost (SIP Type in Communication Use)
GSM VoIP Gateway 4 Port/GoIP Gateway 4 Ports for Call Termination to Save Call Cost (SIP Type in Communication Use)
- 10000 Piece / Pieces per Month
- Shenzhen China
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Product Details
Place of Origin | Guangdong, China | Brand Name | DBL | |
Model Number | GOIP400 | Type | VoIP Gateway |
Product Description
Product name:4 channel GSM VOIP gateway/4 SIM card GOIP gateway/4 port GSM VOIP gateway
General
4 Channel GSM VoIP Gateway is a 4 SIM Card Broadband Phone Gateway.It is a new product that connect the GSM and the VOIP seamlessly. To GOIP 4 channel what is installed on the Mobile SIM Card, you can register the GSM telephone on the VOIP Softswitch.SIP and H.323 agreement are built in the GOIP 4 port and configured flexible.Caller ID can be seen by using SIP. Flexible routing can meet the need of all kinds of call forwarding; even more special is that GOIP 4 support multi-device group, it can be easily combined into arbitrary number of channels of Large Gateway Group.
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 code
•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
Packing(Standard carton,10pcs/carton)
1 pcs of GoIP gateway
1 pcs of adaptor
1 pcs of Ethernet cable
Box Size:35.5*21*6.5cm(L*W*H)
G.W.:0.83kg
Carton Dimension: 43*34*39cm(L*W*H)
G.W.:9.5kg
Warranty: one year
Contact Us
- Shenzhen Etross Telecom Co., Ltd.
- Contact nameFrank Zhou Chat Now
- Phone86-755-28167031
- Address202, 2/f, Block B, Yuetong Comprehensive Bldg., Meilong Road, Longhua Town, Bao'an District, Shenzhen City, Guangdong Province. China, Shenzhen, Guangdong
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