20E1 to SIP Voip Trunk Gateway

20E1 to SIP Voip Trunk Gateway

USD $300 - $400 /Unit

Min.Order:1 Unit

Supply Ability:
100 Unit / Units per Month
Port:
shanghai
Payment Terms:
T/T L/C D/P D/A PayPal
Delivery Detail:
1 days

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Brand Name:
baudcom
Place of Origin:
China

Shanghai Baudcom Communication Device Co.,Ltd

Business Type: Manufacturer
Main Products: 10/100M Managed Media Converter

Product Details

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 InterfacesPSTNSoftware Features
E1/T1 PortsISDN PRILocal/Transparent Ring Back Tone
4/8/12/16/20 E1/T123B+D(T1),30B+D(E1),NT or TE


ITU-T Q.921, ITU-T Q.931, Q.Sig

Overlapping Dialing
DTU Module:Signal 7/SS7Dialing Rules, with up to 2000
4 E1/T1ITU-T, ANSI,ITU-CHINA


MTP1/MTP2/MTP3, TUP/ISUP

PSTN group by E1 port or E1 Timeslot
Interface TypeE1 Frame Type


DF,CRC-4,CRC_ITU

IP Trunk Group Configuration
RJ48(Impedance 120Ω)T1 Frame TypeVoice Codecs Group
Ethernet Interface4-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 EthernetLine CodesCaller and Called Number Black Lists
GE0: 10/100/1000 BaseT Adaptive EthernetE1:NRZ,CMI,AMI,HDB3Access Rule Lists
Serial PortT1:NRZ,CMI,AMI,B8ZSIP Trunk Priority
1* RS232, 115200bpsClockLocal/Remote Clock Source

 

Voice CapabilitiesMaintenanceVoIP 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

EnvironmentalSyslog

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


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