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Dinstar MTG3000-63E1
Rp 303,530,000.00 Excl. PPN
63xE1 Port ISDN Gateway MTG3000-63E1 is a new generation of intelligent voice gateways designed for business / industry / operator business needs. Based on the maintenance, manageable, operational design concept, it has carrier-class, high density, large capacity and other features and IP voice, IP fax, Modem and voice recognition and other enhanced features in one, to provide users with flexible and efficient construction Future-oriented communication network.
Optical port access, high concurrent, high efficiency, carrier-class high reliability
MTG3000 is a high-capacity high-density digital trunking gateway products, carrier-class hardware architecture, 2U height, the maximum support SDH 155M 63E1 (STM-1), dual power supply hot backup, user board hot swap, providing two 155M standard LC SDH optical interface to each other backup, to provide customers with safe, reliable, carrier-class efficient and stable products.
Standard protocol, good compatibility
Support standard SIP protocol, perfect compatibility with IMS / NGN and mainstream softswitch platform
Support PRI / SS7 (ISUP / TUP) signaling, successful docking Huawei, ZTE, Lucent, and so on program-controlled switches, seamless and reliable access operator PSTN network
Recording function
MTG3000 model relay gateway support call recording function, can be full concurrent recording, the use of bypass recording technology, recording services do not affect the normal call, safe and reliable, to ensure that the line operators on the line operation of the security needs, Check, sampling and so on.
Cloud platform management（DMCloud）
DMCloud is a comprehensive network management platform developed by Dingxin. Users can manage the devices deployed in a distributed manner and create different zones and sites according to the actual deployment of the devices to help users master the installation and on-line of the devices. In addition, the system also provides real-time monitoring, alarm push, bill statistics, performance statistics and other functions. Providing remote web, remote telnet, etc., to facilitate remote location, save operation and maintenance costs, improve operation and maintenance efficiency;
SPECIFICATIONS :
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* RS232, 115200bps
PSTN
- • ISDN PRI：
- 23B+D(T1),30B+D(E1),NT or TE ITU-T Q.921, ITU-T Q.931, Q.Sig
- 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
- RTP and Signaling Encryption （VOS RC4)
Voice Capabilities
- Codecs:G.711a/μ law,G.723.1, G.729A/B, iLBC, AMR
- AMR，G.726，G.722，OPUS/SILK（可选）
- 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
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
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
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