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IP Phone Linksys SPA901

IP Phone Linksys SPA901
    
IP Phone Linksys SPA901
CODE: SPA901

Harga: Rp 850.000,00

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Jumlah Kosong

 

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(0356) 324203
0811 3400 840   081 2304 5545
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0856 4941 7122
Sales 3 GAT
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081 2329 1989
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0856 552 533 15
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Comprehensive Interoperability and SIP Based Feature Set
Based on the SIP Standard, the SPA901 has been tested to ensure comprehensive interoperability with equipment from VoIP infrastructure leaders enablind service providers to quickly rollout competitive, feature-rich services to their customers.

 

Linksys SPA901 Entry Level SIP VOIP Phone

With hudreds of features and configurable service parameters, the SPA901 addresses the requirements of traditional business users while leveraging the advantages of IP Telephony. Features such as easy station moves and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA901.

Carrier-Grade Security, Provisioning and Management
The SPA901 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys secure remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high quality support to their subscribers. Remote provisioning also saves service providers the hassle and expense of managing, pre-loading and re-configuring customer premise equipment (CPE).

 

Linksys SPA901 Telephony Features

  • One service provider line
  • Two call appearances accessed via Flash Key or Hook Flash
  • Shared line appearance**
  • Line status indicator
  • Call Hold
  • Music on Hold**
  • Call Waiting
  • Outbound CallerID Blocking
  • Call transfer - Atended and Blind
  • Three Way conferencing with local mixing
  • Multi-Party Call Conferencing via external Conference Bridge**
  • Call Pick Up - Selective and Group**
  • Call park and UnPark**
  • Call back on Busy
  • Call Blocking - Anonymous and Selective
  • Call Forwarding - Unconditional, No Answer, On Busy
  • Call Return - Redial Last Caller
  • Hot Line and Warm Line Automatic Calling
  • Call Logs (60 Entries Each) Made, Answered and Missed Calls. Accessed via HTTP Server
  • Redial Last Called Number
  • Do Not Disturb (Caller Hears Busy Line tone)
  • Block Anonymous Incoming Calls
  • URI (IP) Dialing support (Vanity NUmbers)
  • Built-in Web Server for Administration and Configuration, with User and Admin Access Levels
  • Built-In Interactive Voice Response (IVR) System to check status and change configuration
  • Date and Time w/Intelligent Daylight Savings Support
  • Call Start Time stored in Call Logs
  • Distinctive Ringing
  • 10 User-Downloadable Ring Tones - Ring Tone Generator free from www.Linksys.com
  • Speed Dial (8 entries)
  • Group Paging (Outbound Only)**
  • Intercom (Outbound Only)**
  • Set preferred CODEC, Per Call, All Calls
  • Configurable Dial/Numbering Plan Support
  • Ringer and Handset Voluem Controls
  • DNS SRV and Multiple A Records for Proxy Lookup and Proxy redundancy
  • Syslog, Debug, Report Generation, an Event Logging
  • Secure Call Encrypted Voice Communication Support
  • NAT Traversal
  • Automated Provisioning, Multiple Methods. Up to 256Bit encryption: (HTTP, HTTPS, TFTP)
  • Support Linksys Voice System Automatic Configuration
  • Optionally require Admin Password to Reset unit to factory defaults
  • **Feature requires support by call server.
       

    Hardware

    • Voice Mail Message Waiting Indicator Light
    • Redial Button
    • Dedicated Flash Button
    • Volume Control button cycles through Voluem Levels. Controls Ringer and Handset Volume.
    • Standard 12-Button dialing pad
    • High Quality Handset and Cradle
    • Ethernet LAN - 10Base-T RJ-45
    • 5v DC Universal (100-240v) Switching Power Adapter

    Specifications:

    • Data Networking
      • MAC Address (IEEE 802.3)
      • IPv4
      • ARP
      • DNS
      • DHCP Client
      • ICMP
      • TCP
      • UDP
      • RTP
      • RTCP
      • DiffServ
      • VLAN Tagging
      • SNTP
    • Voice Gateway
      • SIPv2
      • SIP Proxy redundancy
      • Re-Registration with Primary SIP Proxy Server
      • SIP Support in NAT Networks (including STUN)
      • SIPFrag
      • Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP
      • CODEC Name Assignment
      • G.711
      • G.726
      • G.729
      • G.723.1
      • Dynamic Payload Support
      • Adjustable Audio Frames Per Packet
      • DTMF: In-Band and Out-of-Band
      • Flexible Dial Plan Support with Inter-Digit Timers
      • IP Address / URI Dialing Support
      • Call Progress Tone Generation
      • Adaptive Jitter Buffer
      • Frame Los Concealment
      • VAD
      • Attenuation / Gain Adjustments
      • MWI and VMWI
      • Third Party Call Control

    Security

    • Password Protected System, preset to factory default
    • Password Protected access to Administrator and User Level Features
    • HTTPS with Factory Installed Client Certificate
    • HTTP Digest - encrypted authentication via MD5 (RFC 1321)
    • Up to 256-Bit AES Encryption

     


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