20E1 to SIP Voip Trunk Gateway
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Overview BD-20E1-SIP is a new-generation intelligent E1 VoIP gateway which can support 20channels SS…Description
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Description
Overview
BD-20E1-SIP is a new-generation intelligent E1 VoIP gateway which can support 20channels SS7 to SIP or PRI to SIP transmission. The 20E1 voip gateway is ideal application for enterprises, telecom operators and various industries. Focusing on a concept of maintainable, manageable and operable, high integration and large capacity. It provides carrier-grade VoIP and FoIP.services, as well as value-added functions such as modemand voice recognition. Thus it constructs a flexible, high-efficient, future-oriented communication network for users.
BD-20E1-SIP 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 E1 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 E1ports in 1U size
- Support flexible dialing rulesand 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 withPBX of Avaya, NEC and Alcatel, and also leading soft-switch of Huawei,Ciscoand 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: | Software FeaturesLocal/Transparent Ring Back Tone Overlapping Dialing Dialing Rules，with up to 2000 PSTN group by E1 port or E1Timeslot IP Trunk Group Configuration Voice Codecs Group Caller and Called Number White Lists Caller and Called Number IP Trunk Priority |
| 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) , Clock : Local/Remote Clock Source | Voice CapabilitiesCodecs: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 |
| Maintenance Web GUI Configuration Data Backup/Restore PSTN 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 | VoIP ProtocolSIP v2.0 (UDP/TCP),RFC3261 SDP,RTP(RFC2833), RFC3262, 3263,3264,3265,3515,2976,3311SIP TLS/SRTPRTP/RTCP, RFC2198, 1889 SIP-T,RFC3372, RFC3204, RFC3398 SIP Trunk Work Mode : Peer/Access SIP/IMS Registration : With up to 2000 SIP Accounts NAT: Dynamic NAT, Rport |
| Environmental 1+1 Redundancy Power Supply Power Supply: 100-240VAC , 50-60 Hz Power Consumption:45WOperating Temperature:0 ℃ ~ 45 ℃ Storage Temperature Compliance: CE, FCC | Call FeaturesFlexible 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 |
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