Cliente SIP

Turn your phone system into a full functioning SIP/VOIP paging and intercom solution


SIP Client Application is the right software to integrate Barix devices into an existing VOIP phone system. SIP Client works with most SIP based VoIP solutions including Cisco Unified Communications Manager or Asterisk. Using the Session Initiation Protocol (SIP), the client device receives a phone number in addition to its IP address, and can communicate either on a peer-to-peer connection or over a proxy-based connection (PBX). Ringtone and busy tone guarantees that the call remains confidential and the recipient really receives the information. Featuring background music, audio rebroadcasting and special DTMF commands to quickly manage the device, the SIP Client is easy to integrate into an existing telephone system. Four different relay ports can be configured to open doors, windows, or to activate machines while simultaneously speaking.



  • Adding PA functions to existing VOIP telephone systems
  • Adding single button call devices to a VoIP system
  • Cashier’s desk in retail outlets
  • SIP based door phone panels and talk stations
  • SIP addressable loudspeaker
  • SIP based alarm systems on highways, tunnels or railway lines


  • full-duplex phone communication
  • SIP (RFC 3261) compliant architecture
  • Supporting profiles for easy configuration
  • Configurable destination number to call for every input contact
  • Configurable call “pick up”/”hang off” timeout interval
  • Configurable “call/close on level” feature
  • Support for G.711 audio
  • Priority-based notification audio messaging
  • Configurable relay to switch on at call answer/call ring
  • Serial, UDP, TCP or CGI control interface
  • Transparent bidirectional Serial-To-TCP gateway
  • Friendly profile based WEB configuration UI
  • automatic network configuration (BOOTP, DHCP, IPzator)
  • DTMF door open key sequence
  • Barix X8 product support to add 8 additional inputs for predefined calls
  • Features priority notification port