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.
KeyFeatures
•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 BaseTAdaptive Ethernet GE0: 10/100/1000 BaseTAdaptive 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.38and Pass-through Support Modem/POS DTMF Mode: RFC2833/Signal/Inband Clear Channel/Clear Mode
PSTN ISDN PRI: 23B+D(T1),30B+D(E1),NTor TE ITU-T Q.921, ITU-T Q.931, Q.Sig Signal 7/SS7: ITU-T, ANSI,ITU-CHINA MTP1/MTP2/MTP3, TUP/ISUP E1Frame 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 IPTrunk 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 SIPTrunk Work Mode :Peer/Access SIP/IMSRegistration : 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 PSTNCall Statistics SIPTrunk Call Statistics Firmware Upgrade via TFTP/Web SNMP v1/v2/v3 Network Capture Syslog: Debug, Info, Error, Warning , Notice Call History Records via Syslog NTPSynchronization 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