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MTG5000  ISDN VoIP E1 SS7 Media Gateway carrier-grade Digital VoIP gateway
MTG5000  ISDN VoIP E1 SS7 Media Gateway carrier-grade Digital VoIP gateway
MTG5000  ISDN VoIP E1 SS7 Media Gateway carrier-grade Digital VoIP gateway
MTG5000  ISDN VoIP E1 SS7 Media Gateway carrier-grade Digital VoIP gateway

MTG5000 ISDN VoIP E1 SS7 Media Gateway carrier-grade Digital VoIP gateway

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Quick Details

Place of Origin:
Guangdong, China
Channel:
4~64 E1/T1
Interface Type:
RJ48(Impedance 120Ω)
lls Per Second (cps):
130
Protocol:
ISDN PRI, SS7, SIP UDP/TCP
Fax:
Fax over IP (T.38 and Pass-through)
SIP Trunks:
Multiple SIP trunks
Managemet:
DINSTAR Cloud Management
PSR:
Dual Power Supply redundancy
Hot stand-by:
Dual Mainboard hot stand by

Quick Details

Type:
VoIP Gateway
Brand Name:
DINSTAR
Model Number:
MTG5000
Place of Origin:
Guangdong, China
Channel:
4~64 E1/T1
Interface Type:
RJ48(Impedance 120Ω)
lls Per Second (cps):
130
Protocol:
ISDN PRI, SS7, SIP UDP/TCP
Fax:
Fax over IP (T.38 and Pass-through)
SIP Trunks:
Multiple SIP trunks
Managemet:
DINSTAR Cloud Management
PSR:
Dual Power Supply redundancy
Hot stand-by:
Dual Mainboard hot stand by
Products Description

MTG5000 is a new-generation intelligent VoIP gateway, and it is purposely designed for large enterprise network, call center and Telcom service provider to connect with E1/T1 network interfaces. It is developed with the aspect of powerful call control features and maintenance tools. MTG5000 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.
MT5000 supports a range of signaling protocols, offers the connection between SIP and traditional PSTN signaling like SS7 and PRI. It supports multiple codec formats, offers signaling encryption and ASR interface. MTG500 can be implemented with large call center solution, telephone system or IP PBX and service provider for IP/PSTN call services.

MTG Series
Model Options
Description
MTG200
MTG200-1E1
MTG200-2E1
MTG200-4E1

1E1/1T1, 30 Calls
2E1/1T1, 60 Calls
4E1/1T1, 120 Calls on G.711, 60 calls G.723, G.729

MTG1000B
MTG1000B-1E1
MTG1000B-2E1

1E1/1T1, 30 Calls
2E1/1T1, 60 Calls
MTG2000
MTG2000-4E1
MTG2000-8E1
MTG2000-12E1
MTG2000-16E1
MTG2000-20E1

4E1/1T1, 120 Calls
8E1/1T1, 240 Calls
12E1/1T1, 360 Calls
16E1/1T1, 480 Calls
20E1/1T1, 600 Calls
MTG3000
MTG3000-16E1
MTG3000-32E1
MTG3000-48E1
MTG3000-63E1

STM-1, 512Calls
STM-1, 1024 Calls
STM-1, 1536 Calls
STM-1, 2016 Calls

Details Images
key features

Carrier grade hardware design, 1+1 power supply and hardware-based HA Highly scalable and compact design structure, support up to 64 E1 ports(MAX) in 3.5U size Support flexible dialing rules and operations, allowing users to customize dialing rules according to different call routing environments Support multiple codec formats: G.711A/U, G.723.1, G.729A/B and iLBC Very good compatibility, and be workable with Avaya PBX, NEC and Alcatel, and large soft-switch from Huawei, Cisco and ZTE etc.

Project Scenario
Product Paramenters

Physical Interfaces

E1/T1 Ports: 4~64 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: 1* RS232, 115200bps 1* USB2.0

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 : 2-Frame Multi-frame (F12, D3/4) Extended Super-frame (F24, ESF) Line Codes: E1:HDB3 T1: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/In-band Clear Channel/Clear Mode

VoIP Protocol

SIP v2.0 (UDP/TCP),RFC3261 SDP,RTP(RFC2833), RFC3262, 3263,3264,3265,3515,2976,3311 SIP TLS/SRTP RTP/RTCP, RFC2198, 1889 SIP-T,RFC3372, RFC3204, RFC3398 SIP Trunk Work Mode : Peer/Access SIP/IMS Registration :With up to 2000 SIP Accounts NAT: Dynamic NAT, Rport

Call 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 Whitelists Caller and Called Number Blacklists Access Rule Lists IP Trunk Priority

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 Routing Features

Flexible Route Methods PSTN-PSTN, PSTN-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

Environmental

1+1 Redundancy Power Supply Power Supply: 100-240VAC, 50-60 Hz Power Consumption:125W Operating Temperature:0 ℃ ~ 45 ℃ Storage Temperature: -20 ℃ ~80 ℃ Humidity:10%-90% Non-Condensing Dimensions(W/D/H): 437*345*154mm(3.5U) Unit Weight: 12.8kg Compliance: CE, FCC
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