20E1 to SIP Voip Trunk Gateway
Short Description:
Overview BD-MTG-20E1 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, the gateway 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. The VOIP gateway 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…Description
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Description
Overview
BD-MTG-20E1 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, the gateway 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.
The VOIP gateway 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
- Carrier grade hardware design, 1+1 power supply
- High-integrated structure, up to 20 E1 ports in 1U 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.
- Modular design, the capacity can easily adjusted by adding/reduce E1 cards (4E1 per card), reducing project cost.
Physical Interfaces |
PSTN |
Software Features |
E1/T1 Ports |
ISDN PRI |
Local/Transparent Ring Back Tone |
| 4/8/12/16/20 E1/T1 | 23B+D(T1),30B+D(E1),NT or TE
ITU-T Q.921, ITU-T Q.931, Q.Sig |
Overlapping Dialing |
DTU Module: |
Signal 7/SS7 | Dialing Rules, with up to 2000 |
| 4 E1/T1 | ITU-T, ANSI，ITU-CHINA
MTP1/MTP2/MTP3, TUP/ISUP |
PSTN group by E1 port or E1 Timeslot |
Interface Type |
E1 Frame Type
DF,CRC-4,CRC_ITU |
IP Trunk Group Configuration |
| RJ48(Impedance 120Ω） | T1 Frame Type | Voice Codecs Group |
Ethernet Interface |
4-Frame Multi-frame (F4,FT), 2-Frame Multi-frame (F12, D3/4), Extended Super-frame (F24, ESF) , Remote Switch Mode (F72, SLC96) | Caller and Called Number White Lists |
| GE1: 10/100/1000 BaseT Adaptive Ethernet | Line Codes | Caller and Called Number Black Lists |
| GE0: 10/100/1000 BaseT Adaptive Ethernet | E1:NRZ,CMI,AMI,HDB3 | Access Rule Lists |
Serial Port |
T1:NRZ,CMI,AMI,B8ZS | IP Trunk Priority |
| 1* RS232, 115200bps | Clock: Local/Remote Clock Source |
| Voice Capabilities | Maintenance | VoIP Protocol |
| 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/In-band Clear Channel/Clear Mode |
Web GUI Configuration
Data Backup/Restore PSTN Call Statistics SIP Trunk Call Statistics Firmware Upgrade via TFTP/FTP/Web Network Capture SNMP v2 |
SIP v2.0 (UDP/TCP),RFC3261
SDP,RTP(RFC2833), RFC3262, 3263,3264,3265,3515,2976,3311 SIP TLS/SRTP RTP/RTCP, RFC2198, 1889 SIP-T,RFC3372, RFC3204, RFC3398 SIP Trunk Work Mode : Peer/Access |
| Environmental | Syslog |
SIP/IMS Registration With up to 2000 SIP Accounts NAT: Dynamic NAT, Rport |
|
1+1 Redundancy Power Supply Power Supply: 100-240VAC, 50-60 Hz Power Consumption:45W Operating Temperature:0 ℃ ~ 45 ℃ Storage Temperature: -20 ℃ ~80 ℃ Humidity:10%-90% Non-Condensing Dimensions(W/D/H): 436*300*44.5mm(1U) Unit Weight: 3.8kg Compliance: CE, FCC |
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-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 |
Specification
Application
Order information
Q&A
Question1: Can I add static routes in E1 to SIP gateway?
Answer: Yes, it can be added.
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