New Rock - MX120-96FXO [96 FXO Analog VoIP Gateway - 2U Chassis]

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Rp 53.100.000,00


Analog VoIP Gateway with 96-FXO portMX120 is a high capacity and high performance VoIP gateway, packaged in a 2U chassis with up to 96 FXO ports. MX120 is ideal to be deployed by ITSP as gateways to deliver IP-Centrex service in enterprise and residential applications. With dozens of port configuration to choose, it is also ideal to be used in enterprise IP-PBX solutions.

New Rock - MX120-96FXO [96 FXO Analog VoIP Gateway - 2U Chassis]

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Product Overview

MX120 VoIP Getaway is the newest family in Newrock’s MX series gateway products. It offers maximum capacity of 96 FXS lines and/or FXO lines, suitable for carrier and medium enterprise VoIP deployment. Mx120 bridges legacy telecom terminal equipments, such as telephone, FAX machine, enterprise PBX, etc, into IP based voice network that constructed by the carrier or enterprise itself. When placed at network side connecting with PSTN switch, MX120 can be use to bring remote VoIP user into PSTN network. For IP based Call Centersolution, MX120 is the ideal remote access gateway for the operators. MX120 adopts modular design, can flexible configure FXS/FXO line capacity and DSP resource to gain best performance to cost ratio. Mx120 designed with 1U (up to 48 analog lines) or 2U (up to 96 analog lines) high and 19'' wide compact chassis. Full capacity system consumes maximum 125W power, and can be either 110-240V AC or -48V DC input.

Newrock MX120 Analog VoIP Gateway Key Features

  • Applications: IP-centrex, long distance By-pass, IPPBX, call center
  • Flexible hardware configuration, easy system expansion (4 FXS-line or 4 FXO-line incremental)
  • FXO Features: IVR, busy tone/caller ID detection, polarity reversal
  • Support FAX (T.30/T.38), POS machine and modem
  • 500 user configurable routing rules
  • PSTN failover on power failure or network interruption

- Access gateway in IMS based network to deliver IP-centrex serivce
- Media gateway in IP-PBX based enterprise voice network
- PSTN-SIP analog trunking gateway
- Large call center

Intelligent and Rich in Features

Mx120 adopts embedded linux operating system, supports SIP and MGCP. Work as IP-Centrex access terminal, MX120 offers rich features, such as call transfer, call pick-up, conference, caller ID, fashion ring, fork ring, FAX3, Radius call record, 500 routing rule, number translation, PSTN line fail-over on power failure or network interruption, etc. This makes MX120 suitable for variety application requirement. 

Easy to Operate and Maintain

Mx120 provides WEB based management GUI. Through WEB GUI, user can configure gateway parameters, upgrade firmware, download and upload configuration data, monitor status, and etc. MX120 also support other remote management methods, such as Auto Provisioning, Telnet and EMS. 

High Interoperability

Mx120 has passed the interoperability tests with many softswitch and IPPBX suppliers worldwide. It can be deployed by carriers to provide IP-centrex service, and also works with Microsoft OCS 2007, Cisco CallManager, Nortel CS1000 to provide enterprises with Unified Communication solution. 

Investment Protection

When choosing next generation communication equipment for IP network, cost reduction and investment protection are the main challenges. MX120 offers best performance to cost ratio. Through software upgrade, new VoIP function and standard can be continuously delivered to customers. The innovative design in MX120 supports smooth capacity upgrade from 48 lines to 96 lines. This saves customer initial investment, at the same time support feature business grow. 

System Features

SIP protocol
SIP registration (Per trunk, per gateway)
Registration expire setting
SIP trunk
Backup SIP proxy (Up to 10 proxies)
Peer-to-peer communication
SIP-to-SIP relay
Hook flash relay (INFO)

Polarity inverse generation
Caller ID generation (FSK, DTMF, before ring and after ring)
Ring cadence setting
Ring frequency setting
Volume control
Hook flash timing setting
Message waiting indicator (FSK, polarity inverse)

Polarity inverse detection
Caller ID detection (FSK, DTMF, before ring and after ring)
Busy tone detection
DTMF out-pulsing timing setting
Volume control
Ring timing setting
Automatic attendant (2nd ? stage dialing)
Direct inward dialing

G.711ALaw, G.711ULaw, G.729A, G.723.1r63, G.723.1r53, GSM
T.38 fax relay, T.30 fax transparent
Echo cancellation
Dynamic jitter buffer management
Static jitter buffer
DTMF relay (RFC 2833, SIP/INFO, inband)

Voice QoS
IEEE 802.1p tag
DiffServ code point (TOS) bits

Call control
Blind transfer
Explicit transfer
Call forward on busy
Call forward on no answer
Call forward variable
Call waiting
Caller ID
Caller ID blocking
Caller ID on call waiting
Distinctive ring
Do not disturb
Music on hold
Color ring back tone
Call progress tone (Configurable)
Release control (Called party control, calling party control)
Three-way calling
Speed dialing
Calling and called number based routing
Call number transformation (Add, delete, replace)
Hunt group (Sequential and circular hunt)
Ring group
Digit map
PSTN failover (Upon IP network break or failure to reach SIP proxy, or power break)

NAT traversal (STUN)

IP access list (IP table)
SIP/RTP/TELNET/HTTP/TFTP port assignment
Web-utility access privilege (Admin and user)

System management
TR069-based management (Include TR069, TR104, and TR106)
Standard SNMP agent (MIB v2)
Web-based management interface (Local and remote access)
Firmware upgrade
Log management (8 levels)
Various debugging and call trace
TCP dump
Remote management with console and telnet
Configuration files import and export
System status monitoring and statistics



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