MTG2000 is a new-generation intelligent VoIP gateway, which is designed for enterprises, telecom operators and various industries. Focusing on a concept of maintainable, manageable and operable, MTG2000 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.
MTG2000 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 access networks of SMEs, call centers, telecom operators and large-scale enterprises.
Key Features:
Multi-port and high-integrated structure; up to 20 E1/T1 with 1U size
Provide various services, such as VoIP, FoIP, Modem and POS
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:
E1/T1 Ports
4/8/12/16/20 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
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 :
4-Frame Multi-frame (F4,FT),2-Frame Multi-frame (F12, D3/4),Extended Super-frame (F24, ESF) ,Remote Switch Mode (F72, SLC96)
Line Codes:
E1:NRZ,CMI,AMI,HDB3
T1:NRZ,CMI,AMI,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/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