Audited Supplier
MTG3000
Gateway
TCP/IP
SIP
64
Wire
Black
16/32/48/63
up to 1890
Unlimited
Yes
Yes
Yes
Yes
Yes
Yes
DINSTAR
Gift Box
CE, ISO90001
China
851762
Product Description
High Capacity Digital VoIP E1/T1 Gateway MTG3000 , up to 63 E1s, For Service Providers & Telecom Operators
MTG3000 is a carrier grade VoIP gateway, which is designed for telecom operators, ITSPs with high reliability and performance. Focusing on a concept of maintainable, manageable and operable, MTG3000 adopts STM-1 interface which 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.
MTG3000 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 networks of ITSPs, telecom operators and large-scale enterprises.
Key Features
Physical Interfaces
SDH Interfaces
2* Standard LC SDH, 155M
1+1 Redundancy Channels Protection
Master/Slave Clock Source
Main Control Unit(MCU)
1+1 Redundancy, Hot Plug
Digital Processing Unit (DTU)
4* DTU Maximum
Support 512 Voice Channels Each Board
Ethernet Interface
GE1: 10/100/1000 BaseT Adaptive Ethernet
GE0: 10/100/1000 BaseT Adaptive Ethernet
Serial Port
1* RS232, 115200bps
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
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,MF_CRC,MF
Line Code: HDB3
Clock Source:
Local/Remote Clock Source
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
VoIP Protocol
SIP v2.0 (UDP/TCP),RFC3261
SDP,RTP(RFC2833), RFC3262,
3263,3264,3265,3515,2976,3311
RTP/RTCP, RFC2198, 1889
TLS/SRTP
SIP-T,RFC3372, RFC3204, RFC3398
SIP Trunk Work Mode :Peer/Access
SIP/IMS Registration :
with up to 256 SIP Accounts
NAT: Dynamic NAT, Rport
Call Feature
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
Maintenance
Web GUI Configuration, HTTP/HTTPS
Data Backup/Restore
PSTN Call Statistics
SIP Trunk Call Statistics
Firmware Upgrade via TFTP/Web
SNMP v1/v2/v3
Network Capture
Syslog:
Debug, Info, Error, Warning , Notice
Call History Records via Syslog
NTP Synchronization
Centralized Management System
Environmental
Redundant Power Supply
Power Supply: 100-240VAC, 50-60 Hz
Power Consumption:110W
Operating Temperature:0 ºC ~ 45 ºC
Storage Temperature: -20 ºC ~80 ºC
Humidity:10%-90% Non-Condensing
Dimensions(W/D/H): 437*320*88mm(2U)
Unit Weight: 6.5kg
Compliance: CE, FCC,CCC
MTG3000 is a carrier grade VoIP gateway, which is designed for telecom operators, ITSPs with high reliability and performance. Focusing on a concept of maintainable, manageable and operable, MTG3000 adopts STM-1 interface which 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.
MTG3000 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 networks of ITSPs, telecom operators and large-scale enterprises.
Key Features
•Carrier grade hardware design, 1+1 power supply and MCU, hot plug
•High-integrated structure, STM-1 155M (63*E1) in 2U size
•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
SDH Interfaces
2* Standard LC SDH, 155M
1+1 Redundancy Channels Protection
Master/Slave Clock Source
Main Control Unit(MCU)
1+1 Redundancy, Hot Plug
Digital Processing Unit (DTU)
4* DTU Maximum
Support 512 Voice Channels Each Board
Ethernet Interface
GE1: 10/100/1000 BaseT Adaptive Ethernet
GE0: 10/100/1000 BaseT Adaptive Ethernet
Serial Port
1* RS232, 115200bps
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
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,MF_CRC,MF
Line Code: HDB3
Clock Source:
Local/Remote Clock Source
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
VoIP Protocol
SIP v2.0 (UDP/TCP),RFC3261
SDP,RTP(RFC2833), RFC3262,
3263,3264,3265,3515,2976,3311
RTP/RTCP, RFC2198, 1889
TLS/SRTP
SIP-T,RFC3372, RFC3204, RFC3398
SIP Trunk Work Mode :Peer/Access
SIP/IMS Registration :
with up to 256 SIP Accounts
NAT: Dynamic NAT, Rport
Call Feature
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
Maintenance
Web GUI Configuration, HTTP/HTTPS
Data Backup/Restore
PSTN Call Statistics
SIP Trunk Call Statistics
Firmware Upgrade via TFTP/Web
SNMP v1/v2/v3
Network Capture
Syslog:
Debug, Info, Error, Warning , Notice
Call History Records via Syslog
NTP Synchronization
Centralized Management System
Environmental
Redundant Power Supply
Power Supply: 100-240VAC, 50-60 Hz
Power Consumption:110W
Operating Temperature:0 ºC ~ 45 ºC
Storage Temperature: -20 ºC ~80 ºC
Humidity:10%-90% Non-Condensing
Dimensions(W/D/H): 437*320*88mm(2U)
Unit Weight: 6.5kg
Compliance: CE, FCC,CCC
