MTG3000 High Density Digital VoIP Gateway

MTG3000-16E1  --  16*E1s 2U chassis, SIP, PRI/SS7 G.711A/U,G.723.1,G.729A/B, iLBC,AMR, SDH STM-1 155M interface 1+1 Redundant power supplies
MTG3000-32E1  --  32*E1s 2U chassis, SIP, PRI/SS7 G.711A/U,G.723.1,G.729A/B, iLBC,AMR, SDH STM-1 155M interface 1+1 Redundant power supplies
MTG3000-48E1  --  48*E1s 2U chassis, SIP, PRI/SS7 G.711A/U,G.723.1,G.729A/B, iLBC,AMR, SDH STM-1 155M interface 1+1 Redundant power supplies
MTG3000-63E1  --  63*E1s 2U chassis, SIP, PRI/SS7 G.711A/U,G.723.1,G.729A/B, iLBC,AMR, SDH STM-1 155M interface 1+1 Redundant power supplies

License of 1+1 Redundant MCU Mainboards (Optional)

MTG3000 supports a wide-range of signaling protocols, realizing the interconnection between SIP and traditional signals like ISDN PRI / SS7, utilizing efficiency of trunking resources while ensuring voice quality. With multiple voice codes, secure signal encryption and smart voice recognition technology, MTG3000 is ideal for a variety of applications of services providers and telecom operators.

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High Capacity Digital VoIP Gateway for Carriers & ITSPs

• 16 to 63 ports E1/T1 in 2U chassis, STM-1 interface
• Up to 1890 simultaneous calls
• Redundancy Dual MCU units
• Dual Power Supplies
• Flexible routing 
• Multiple SIP trunks
• Fully compatible with mainstream VoIP platforms

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Rich Experiences on PSTN Protocols

• ISDN PRI
• ISDN SS7, SS7 links redundancy
• R2 MFC
• T.38,Pass-through fax, 
• Support modem and POS machines
• More than 10-year expriences to integrate with a wide range of Legacy PBXs / Service providers' PSTN networks

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Features:

• 1+1 Redundant Main Control Unit (MCU)
• Up to 63 E1s/T1s, STM-1 interface
• 4 Digital Processing Unit (DTU), each support 512 channels
• Codecs:G.711a/μ law,G.723.1, G.729A/B, iLBC 13k/15k,AMR
• Dual Power Supplies
• Silence Suppression
• 2 GE
• Comfort Noise
• SIP v2.0
• Voice Activity Detection
• SIP-T,RFC3372, RFC3204, RFC3398
• Echo Cancellation (G.168),with up to 128ms
• SIP Trunk Work Mode: Peer/Access
• Adaptive Dynamic Buffer
• SIP/IMS Registration :with up to 256 SIP Accounts
• Voice, Fax Gain Control
• NAT: Dynamic NAT, Rport
• FAX:T.38 and Pass-through
• Flexible Route Methods: PSTN-PSTN, PSTN-IP, IP-PSTN
• Support Modem/POS
• Intelligent Routing Rules
• DTMF Mode: RFC2833/SIP Info/In-band
• Call Routing base on Time
• Clear Channel/Clear Mode
• Call Routing base on Caller/Called Prefixes
• ISDN PRI:
• 256 Route Rules for each Direction
• Signal 7/SS7: ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP
• Caller and Called Number Manipulation
• R2 MFC
• Local/Transparent Ring Back Tone
• Web GUI Configuration
• Overlapping Dialing
• Data Backup/Restore
• Dialing Rules, with up to 2000
• PSTN Call Statistics
• PSTN group by E1 port or E1 Timeslot
• SIP Trunk Call Statistics
• IP Trunk Group Configuration
• Firmware Upgrade via TFTP/Web
• Voice Codecs Group
• SNMP v1/v2/v3
• Caller and Called Number White Lists
• Network Capture
• Caller and Called Number Black Lists
• Syslog: Debug, Info, Error, Warning , Notice
• Access Rule Lists
• Call History Records via Syslog
• IP Trunk Priority
• NTP Synchronization
• Radius
• Centralized Management System