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SHENZHEN DINSTAR CO., LTD.
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ChinaVIP0Audited Supplier
Products: VoIP Phone , GSM Gateway , VoIP Gateway , IP Phone , FXS VoIP Gateway , FXO VoIP Gateway , E1 Gateway , VoIP , Wireless , Asterisk
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interoperable with PBX of Avaya, NEC and Alcatel, Dinstar 4/8/20 channels SS7 media gateway
OverviewQuick DetailsPlace of Origin:Guangdong, China (Mainland)Brand Name:DinstarModel Number:MTG2000-20E1Type:VoIP GatewayDialing Rules,:w
$100 /
Stock: 10000 s
Quantity:
-+
Min Order: piece
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Overview
Quick Details
Place of Origin:
Guangdong, China (Mainland)
Brand Name:
Dinstar
Model Number:
MTG2000-20E1
Type:
VoIP Gateway
Dialing Rules,:
with up to 2000
Syslog::
Debug, Info, Error, Warning , Notice
E1 Frame Type ::
DF,CRC-4,CRC_ITU
COLOR:
BLACK
Supply Ability
Supply Ability:
500 Unit/Units per Month
Packaging & Delivery
Packaging Details
CARTON
Port
HK/SHENZHEN

interoperable with PBX of Avaya, NEC and Alcatel, Dinstar 4/8/20 channels SS7 media gateway

 MTG2000 is a new-generation intelligent VoIP gateway, which is designed for enterprises, telecom operators and various industries. Focusing on a concept of maintainable, manageable and operable, MTG2000 features high integration and large capacity. It provides carrier-grade VoIP and FoIP . services, as well as value-added functions such as modem and voice recognition. Thus it constructs a flexible, high-efficient, future-oriented communication network for users.

 

MTG2000 supports a range of signaling protocols, realizing the interconnection between SIP and traditional signals like SS7 and PRI. It supports multiple codec methods, offers signal encryption technology and smart voice recognition technology, and improves the utilizing efficiency of trucking resources while ensuring voice quality. The trunk gateway is ideally fit for various access networks of SMEs, call centers, telecom operators and large-scale enterprises.

Key Features:

Multi-port and high-integrated structure; up to 20 E1/T1 with 1U size
Provide various services, such as VoIP, FoIP, Modem and POS

Support flexible dialing rules and operations, allowing users to customize dialing rules according to different working environments

Support multiple coding standards: G.711A/U, G.723.1, G.729A/B and iLBC

 

High compatibility, interoperable with PBX of Avaya, NEC and Alcatel, and also leading soft-switch of Huawei,Cisco and ZTE etc.

Physical Interfaces:

E1/T1 Ports

     4/8/12/16/20 E1/T1

DTU Module :

     4 E1/T1

Interface Type

     RJ48(Impedance 120Ω)

Ethernet Interface

    GE1: 10/100/1000 BaseT Adaptive Ethernet

    GE0: 10/100/1000 BaseT Adaptive Ethernet

Serial Port

    1RS232, 115200bps 

PSTN:

ISDN PRI

    23B+D(T1),30B+D(E1),NT or TE

    ITU-T Q.921, ITU-T Q.931, Q.Sig

Signal 7/SS7

     ITU-T, ANSI,ITU-CHINA

     MTP1/MTP2/MTP3, TUP/ISUP

E1 Frame Type : DF,CRC-4,CRC_ITU

T1 Frame Type :

     4-Frame Multi-frame (F4,FT),2-Frame  Multi-frame (F12, D3/4),Extended Super-frame (F24, ESF) ,Remote Switch Mode (F72, SLC96)

Line Codes:

    E1:NRZ,CMI,AMI,HDB3

    T1:NRZ,CMI,AMI,B8ZS

Clock

     Local/Remote Clock Source

Voice Capabilities:

Codecs:G.711a/μ law,G.723.1, G.729A/B, iLBC, AMR
Silence Suppression
Comfort Noise
Voice Activity Detection
Echo Cancellation (G.168),with up to 128ms
Adaptive Dynamic Buffer
Voice ,Fax Gain Control 
 FAX:T.38 and Pass-through
Support Modem/POS
DTMF Mode: RFC2833/Signal/Inband

Clear Channel/Clear Mode

VoIP Protocol:

SIP v2.0 (UDP/TCP),RFC3261

     SDP,RTP(RFC2833), RFC3262,

     3263,3264,3265,3515,2976,3311

RTP/RTCP, RFC2198, 1889

SIP-T,RFC3372, RFC3204, RFC3398

SIP Trunk Work Mode :Peer/Access

SIP/IMS Registration :with up to 256 SIP Accounts

NAT: Dynamic NAT, Rport

Software Features:

Local/Transparent Ring Back Tone
Overlapping Dialing
Dialing Rules,with up to 2000
PSTN group by E1 port or E1 Timeslot
IP Trunk Group Configuration
Voice Codecs Group
Caller and Called Number White Lists
Caller and Called Number Black Lists
Access Rule Lists
IP Trunk Priority

RTP and Signaling Encryption (VOS RC4)

Maintenance:

Web GUI Configuration
Data Backup/Restore 
PSTN Call Statistics
SIP Trunk Call Statistics 
Firmware Upgrade via TFTP/FTP/Web
Network Capture
SNMP v2
Syslog: Debug, Info, Error, Warning , Notice
Call History Records via Syslog
NTP Synchronization

Centralized Management System

Call Features:

Flexible Route Methods

   PSTN-PSTN, PSTN-IP, IP-IP, IP-PSTN

Intelligent Routing Rules
Call Routing base on Time
Call Routing base on Caller/Called Prefixes
256 Route Rules for each Direction

Caller and Called Number Manipulation 

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