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High Density Digital VoIP Gateway Model JSLTG5000

High Density Digital VoIP Gateway Model JSLTG5000

Short Description:

JSLTG5000 is a carrier-grade Digital VoIP gateway, and it is purposely designed for large enterprise networks, call centers, and Telecom service providers to connect with E1/T1 network interfaces. It is developed with the aspect of powerful call control features and maintenance tools. JSLTG5000 supports high density calls with a very stable system support. It also provides carrier grade VoIP and FoIP services, as well as value added functions such as fax modem and voice recognition service.

 


Product Detail

Product Tags

JSLTG5000

JSLTG5000 is a carrier-grade Digital VoIP gateway, and it is purposely designed for large enterprise networks, call centers, and Telecom service providers to connect with E1/T1 network interfaces. It is developed with the aspect of powerful call control features and maintenance tools. JSLTG5000 supports high density calls with a very stable system support. It also provides carrier grade VoIP and FoIP services, as well as value added functions such as fax modem and voice recognition service.

 

Product Fetures

•64 E1/T1 Ports

•4 Digital Processing Unit (DTU), each support 480 channels

•Codecs: G.711A/U, G.723.1, G.729A/B and iLBC

•Dual Power Supplies

•Silence Suppression

•2 GE

•Comfort Noise

•SIP v2.0

•Voice Activity Detection

•SIP-T, RFC3372, RFC3204, RFC3398

•Echo Cancellation (G.168), with up to 128ms

•SIP Trunk Work Mode: Peer/Access

•Adaptive Dynamic Buffer

•SIP/IMS Registration: with up to 2000 SIP Accounts

•Voice, Fax Gain Control

•NAT: Dynamic NAT, Rport

•FAX: T.38 and Pass-through

•Flexible Route Methods: PSTN-PSTN, PSTN-IP, IP-PSTN

•Support Modem/POS

•Intelligent Routing Rules

•DTMF Mode: RFC2833/SIP Info/In-band

•Call Routing base on Time

•Clear Channel/Clear Mode

•Call Routing base on Caller/Called Prefixes

•ISDN PRI

•512 Route Rules for each Direction

•Signal 7/SS7: ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP

•Caller and Called Number Manipulation

•R2 MFC

•Local/Transparent Ring Back Tone

•Web GUI Configuration

•Overlapping Dialing

•Data Backup/Restore

•Dialing Rules, with up to 2000

•PSTN Call Statistics

•PSTN group by E1 port or E1 Timeslot

•SIP Trunk Call Statistics

•IP Trunk Group Configuration

•Firmware Upgrade via TFTP/Web

•Voice Codecs Group

•SNMP v1/v2/v3

•Caller and Called Number White Lists

•Network Capture

•Caller and Called Number Black Lists

•Syslog: Debug, Info, Error, Warning , Notice

•Access Rule Lists

•Call History Records via Syslog

•IP Trunk Priority

•NTP Synchronization

•Radius

•Centralized Management System

 

Product detail

High Capacity Digital VoIP Gateway for Carriers & ITSPs

64  E1/T1 ports

Up to 1920 simultaneous calls

Dual Power Supplies

Flexible routing

Multiple SIP trunks

Fully compatible with mainstream VoIP platforms

0a-01

Rich Experiences on PSTN Protocols

ISDN PRI

 

ISDN SS7, SS7 links redundancy

R2 MFC

T.38, Pass-through fax,

Support modem and POS machines

More than 10-year expriences to integrate with a wide range of Legacy PBXs / Service providers' PSTN networks

dxj1-2
E1-T1

E1/T1

T.38

T.38/T.30

PRI-

PRI

SS7-

SS7

NGN-IMS

NGN/IMS

SNMP-

SNMP

Easy Management

Intuitive Web interface

Support SNMP

Automated provisioning

CASHLY Cloud Management System

Configuration Backup & Restore

Advanced Debug tools

MTG200

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