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8 GSM SIM Channel GoIP Gateway TGG800

China

TGG800

650 ~ 880 / Piece ( Negotiable ) Get Latest Price

China

650 per Piece

1 Piece

200mm (W) x 370mm (L) x 72mm (H) 1.5KG

4

100 Carton per Week

T/T, L/C, D/A, D/P, Western Union, PayPal

-

今すぐお問い合わせください

製品仕様

GSM GATEWAY : VOIP

タイトルまたは説明

VoIP GSM Gateway TGG800 are VoIP GSM Gateway for call termination (VoIP to GSM) and origination (GSM to VoIP). It is SIP&H.323 based and compatible with Asterisk, Trixbox, 3CX, SIP Proxy Server, VoipBuster. It can enable to make 8 calls simultaneously from IP phones to GSM networks and GSM networks to IP phone.

Key Features

8 GSM channels, up to 8 SIM cards
For call termination (VoIP to GSM) and origination (GSM to VoIP)
Standard SIP & H.323 protocol, Communicates with other gateway or PC
Quad band, IMEI changeable, Remote Access
Support of SMB16 SIM Bank
Optional SMS termination
Allows your program send/receive SMS with AT command
Easy to install and administrate
Auto Balance and Recharge
Auto BTS changeable

Support one stage dialing
Support free mode-two stage dialing and assigned mode-one stage dialing
Call Back feature
All functions can be set on web
Provide CDR
LEDs for Power, Ready, Status, WAN, PC, GSM
Dial in mode or dial out mode only
Call forward from GSM to VoIP and VoIP to GSM
Dial Plan
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal
Dynamic selection of codec


Advanced Features


Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
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


Network Features


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 (STUN)- Simple Traversal of 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
Web-base Management
PPP over Ethernet (PPPoE)
PPP Authentication Protocol (PAP)
Internet Control Message Protocol (ICMP)
TFTP Client................................

お問い合わせ

連絡先担当者: Leo

会社:TRIXAS TECHNOLOGY CO., LTD

住所:GAOXIN, xian, Shanxi, China,

To,

Leo < TRIXAS TECHNOLOGY CO., LTD >
私は知りたい:

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