PRODUCT SPECIFICATIONS
DINSTAR MTG200 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 MTG200 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.
MTG200 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
- Up to 60 simultaneous calls with 2x E1/T1 ports
- Support echo cancellation, DJB, CNG, VAD and QoS
- Use of existing E1/T1 ISDN PRI interface of the PBX
- Use of existing VoIP interface of the IP-PBX
- Maintain existing dialing habits and business communication patterns
- Built-in Web-based management, SNMP, command line interface (CLI)
- Standard ISDN compliant and Interoperable with a wide range of IP-PBXs
Dimensions(W/D/H)
- 225 x 150 x 38mm (1U)
Unit Weight
- 0.8kg
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
- SS7 (optional)
- ITU-T, ANSIļ¼ITU-CHINA
- MTP1/MTP2/MTP3, TUP/ISUP
- R2 MFC (optional)
- 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
- 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
- VLAN 802.1p/q
- 60 Concurrent Calls
- 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 100-240VAC@DC12V1A
- 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
- Voice Codecs Group
- Access Rule Lists
- 100 SIP Trunks
- Route direction: PSTN-IP, IP-PSTN, PSTN-PSTN, IP-IP
- Maintenance
- Web GUI Configuration, HTTP/HTTPS
- 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