Dinstar MTG3000 16 E1-T1 Trunk Gateway (MTG3000-16E1)

Dinstar MTG3000 16 E1-T1 Trunk Gateway (MTG3000-16E1) The Dinstar MTG3000 E1-T1 Trunk Gateway 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. Dinstar MTG3000 Key Features Carrier grade hardware design, 1+1 power supply and MCU, hot plug High-integrated structure, STM-1 155M (16*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 softswitch of Huawei, Cisco and ZTE etc. Dinstar MTG3000 - Technical 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 1* RS232, 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,MF_CRC,MF Line Code HDB3 Clock Source 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 Environmental Redundant Power Supply Power Supply: 100-240VAC, 50-60 Hz Power Consumption:110W Operating Temperature:0 ~ 45 Storage Temperature: -20 ~80 Humidity:10percent-90percent Non-Condensing Dimensions(W-D-H): 437*320*88mm(2U) Unit Weight: 6.5kg Compliance: CE, FCC,CCC