WSS120 series VoIP Access gateway are a telecom carrier class VoIP Gateway designed for regular phone service, Fax service and IP Centrex service,WSS120 can be used by carrier and enterprise as part of the cost effective IP-PBX solutions;

WSS120 is stackable platform. The main unit can work stand along with capability of maximum 48 analog lines. When hook to an expansion unit, it can provide as many as 96 analog lines while still managed as a single GW; WSS120 main unit and expansion unit have the same physical size of 44x440x300 mm (Height x Width x Depth), the two units are connected through a 36 pin ribbon cable to form a WSS120 system; The main unit provides two Ethernet ports (RJ-45), featuring 10/100M auto negotiation and auto crossover adjustment. The main unit and expansion unit are separated powered through 100-240 AC, 50/60HZ. The maximum power consumption is 70W for main unit and 55W for the expansion unit.

Main Applications

• IPCentrex,Remote SIP access gateway for IP-PBX like Asterisk, 3CX...
• IP based wide area PBX for enterprise, government branches and schools 
• VoIP Long Distance application for IP phone supermarkets and enterprises 
  Carrier primary and secondary line service 
• 48,96FXO ports for calling card, call termination with PSTN lines or GSM/CDMA terminal
• PBX functions, like IVR, call transfer, Call hold...

Part# RealTone WSS100-TG models:

• General
• Features
• Protocols
• Call Features
• Documents

RealTone WSS120 Gateway hỗ trợ 48-96 FXS/FXO Port
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• MGCP or SIP access gateways 
• SIP-PSTN analog gateway 
• Routing (based on called number or other criteria) 
• Radius based CDR for accounting 
• STUN client for NAT transversal 
• Flexible interface for analog phones, fax, traditional PBX with analog trunks. 
  Redundant power supply option 
• Hot swappable modules 
• Fax tone detection and codec bypass (T.30) and fax relay (T.38) function 
• G.711, G.723.1, G.729A, iLBC, GSM codec 
• G.168 echo cancellation 
• DTMF transportation: transparent mode, RFC 2833 mode, SIP/INFO mode 
• Fashion Color ringing, MWI by Tone or Neon light, Routing table, Speed dial

• Session Initiated Protocol (SIP) 
• Media Gateway Control Protocol (MGCP) 
• Real Time Transport Protocol(RTP/RTCP) 
• TFTP for remote software upgrade 
• HTTP for system and interface configuration and monitoring 
• SNMP v.1, v.2, v.3 
• Telnet for remote support 
• DHCP for dynamic IP address allocation 
• DTMF 
• STUN Client for NAT/firewall traverse

• Caller ID
• Message waiting indication 
• Call waiting 
• Call forwarding 
• Caller id on call waiting 
• Call transfer (on busy, no answer, all) 
• Special purpose calling codes 
• Speed dial 
• Redials

- Brochure
- Datasheet

- Quick Start
- User Manual