New Rock - MX60-32S [32 FXS Analog VoIP Gateway]

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Rp 18.000.000,00


Analog VoIP Gateway with 32-FXS port, MX60 is the successor of MX51 VoIP gateway, delivering more processing power in smaller footprint. A MX60 provides up to 48 ports of FXS and FXO and 2 Ethernet ports. It is a high cost-effective solution for wide range of enterprise VoIP applications.

New Rock - MX60-32S [32 FXS Analog VoIP Gateway]

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Product Overview

MX60 VoIP Gateway is a member of MX series developed by New Rock Technologies, Inc., which is designed for multi-purposes applications. Supporting SIP and MGCP for call signaling and TR069 for management as well, MX60 enables vast deployment in delivering carrier-hosted converged services as well as enterprise-based voice applications. A MX60 gateway is typically used to connect analog telephone terminals, PBXs or key systems to the IP network through FXO or FXS ports, and is ideal components in many VoIP-based solutions. 

Newrock MX60 VoIP Gateway Key Features

  • Support 3GPP IMS
  • Support TR069/TR104/TR106 for remote management
  • Flexible configuration of FXS/FXO ports
  • PSTN failover on power failure or network interruption
  • 500 routing rule capacity
  • IP filter, encryption for security
  • Support Fax (T.30/T.38), POS machine and modem
  • Busy tone detection and polarity reversal of FXO ports
  • Compatible with unified communication solutions, such as CallManager, OCS, and Asterisk 
Intelligent and Rich in Features

In a highly compact 1U chassis, MX60 adopts embedded Linux operating system and offers rich features, such as call transfer, call pick-up, built-in 3-way conference, caller ID, CRBT, T.38 Fax relay, flexible call routing with 500 rule capacity, number translation, PSTN failover on power failure or network interruption, and etc. MX60 is powered by high speed CPU and dedicated DSP chip sets, which allow MX60 to be used in highest call traffic applications. 

Easy to Operate and Maintain

MX60 provides Web-based management GUI, allowing user to configure parameters, upgrade firmware, import and export configuration data, monitor status, and etc. MX60 also supports remote management standards, such as Auto-provisioning, Telnet, TFTP, SNMPv2, TR069, TR104 and TR106. 

High Interoperability

MX60 has performed the interoperability tests with many softswitch and IPPBX from worldwide vendors, including IP-PBX from Microsoft (OCS), Cisco (CallManager), Nortel (CS1000), and softswitch from Huawei, ZTE, Ericson, and etc. 

Free Upgrade

When choosing next generation communication equipment for IP network, cost reduction and investment protection are the main challenges. MX60 offers best performance of cost reduction. Through software upgrade, latest VoIP functions and standard can be continuously delivered to customers and prolong cycle time of the equipment.

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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