PRODUCT SPECIFICATIONS
DINSTAR MTG1000 Trunk Gateway enables direct routing of calls between the fixed line ISDN and the cost effective IP networks for capitalizing on low cost VoIP telephony. By integrating MTG1000 Trunk Gateway with existing PBX / PABX telephone systems, businesses of all sizes can benefit from low-cost Voice over IP calls and achieve substantial cost savings without the high upfront costs associated with communications infrastructure changes related with the migration to VoIP.
MTG1000 offering PBX/IP-PBX users a powerful telephony cost saving device and advanced flexible routing capabilities between ISDN PRI/SS7 to SIP, and also can be situated between the PBX and the PSTN, saving the cost of an additional PBX E1/T1 port. It is ideal for telecom distributors, resellers and service providers.
Key Features
- Efficient concurrent processing, up to 60 channels
- Support flexible dialing rules and operations, allowing users to customize dialing rules according to different working environments
- Support T.38,Pass-through fax, as well as modem and POS machines
- Support multiple coding standards: G.711A/U, G.723.1, G.729A/B and AMR
- High compatibility, interoperable with PBX of Avaya, NEC and Alcatel, and also leading soft-switch of Huawei, Cisco and ZTE etc.
Dimensions(W/D/H)
- 436 x 300 x 44.5mm (1U)
Unit Weight
- 2.0kg
Technical Description
- Physical Interfaces
- E1/T1 Ports
- 1 to 2 E1/T1
- Interface Type
- RJ48(Impedance 120Ωļ¼
- Ethernet Interface
- GE1: 10/100/1000 Base-T Adaptive Ethernet
- GE0: 10/100/1000 Base-T Adaptive Ethernet
- Serial Port
- 1x RS232, 115200bps
- E1/T1 Ports
- 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, ANSIļ¼ITU-CHINA
- MTP1/MTP2/MTP3, TUP/ISUP
- R2 MFC
- China and other 22 variants standard
- 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
- ISDN PRI
- Call Features
- 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
- Flexible Route Methods
- Voice Capabilities
- Codecs
- G.711a/μ law,G.723.1,
- G.729A/B, iLBC 13k/15k,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/SIP Info/In-band
- Clear Channel/Clear Mode
- Codecs
- VoIP Protocol
- SIP v2.0 (UDP/TCP),RFC3261
- SDP,RTP(RFC2833), RFC3262, 3263,3264,3265,3515,2976,3311
- RTP/RTCP, RFC2198, 1889
- SIP TLS/SRTP
- SIP-T, RFC3372, RFC3204, RFC3398
- SIP Trunk Work Mode :Peer/Access
- SIP/IMS Registration
- With up to 256 SIP Accounts
- NAT: Dynamic NAT, Rport
- Environmental
- 1+1 Redundancy Power Supply
- Input 100-240 VAC, 50-60 Hz
- Power Consumption: 15W
- Operating Temperature: 0 °C ~ 45 °C
- Storage Temperature: -20 °C ~80 °C
- Humidity:10%-90% Non-Condensing
- Compliance: CE, FCC
- 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
- Maintenance
- Web GUI Configuration
- Data Backup/Restore
- PSTN Call Statistics
- SIP Trunk Call Statistics
- Firmware Upgrade via TFTP/Web
- SNMP v1/v2/v3
- Network Capture
- Syslog:
- Debug, Info, Error, Warning , Notice
- Call History Records via Syslog
- NTP Synchronization
- Centralized Management System