COMPATIBILE CON ASTERISK

Basato sul protocollo SIP, lo SPA941 è testato per assicurare piena interoperabilità con le più disparate realtà e strutture VoIP, compreso la soluzione IP PBX Asterisk. Presenta un ampio display ad alta risoluzione, speaker, jack cuffie da 2.5mm e una porta Ethernet RJ45 a 10Mbps. Supporta due lenee, trasferimento chiamata e la possibilità di mettere una chiamata in attesa per risporde ad un'altra. Tramite aggiornamenton software è possibilie portare le linee fino a 4. Ogni linea può essere configurata su un unico numero telefonico o estensione (extension), o su una serie di numeri condivisi fra più apparecchi telefonici.
Da segnalare la possibilità di usare protocolli di criptaggio standard che assicurano un sistema di provisioning e aggiornamenti da remoto sicuro e affidabile.  

Protocollo SIPv2 Supporto STU

Dettagli tecnici

 Up to Four Call Appearances with Independent Configuration and Registration
     - The SPA941 ships with two line appearances enabled. A two line upgrade is     available via a software license key
    installed locally using the SPA941 web interface, or installed remotely via a     secure profile update.
 Pixel Based Display: 128x64 Monochrome Graphical Liquid Crystal Display (LCD)
 Line Status - Active Line Indication, Name and Number
 Menu Driven User Interface
 Digits Dialed with Number Auto-Completion
 Shared Line Appearance **
 Speakerphone
 Call Hold
 Music on Hold **
 Call Waiting
 Caller ID Name and Number and Outbound Caller ID Blocking
 Outbound Caller ID Blocking
 Call Transfer - Attended and Blind
 Call Conferencing
 Automatic Redial
 On-Hook Dialing
 Call Pick Up - Selective and Group **
 Call Park and UnPark **
 Call Swap
 Call Back on Busy
 Call Blocking - Anonymous and Selective
 Call Forwarding - Unconditional, No Answer, On Busy
 Hot Line and Warm Line Automatic Calling
 Call Logs (60 entries each): Made, Answered, and Missed Calls
 Redial from Call Logs
 Personal Directory with Auto-dial (100 entries)
 Do Not Disturb (callers hear line busy tone)
 URI (IP) Dialing Support (Vanity Numbers)
 On Hook Default Audio Configuration (Speakerphone and Headset)
 Multiple Ring Tones with Selectable Ring Tone per Line
 Called Number with Directory Name Matching
 Call Number using Name - Directory Matching or via Caller ID
 Subsequent Incoming Calls with Calling Name and Number
 Date and Time with Intelligent Daylight Savings Support
 Call Duration and Start Time Stored in Call Logs
 Call Timer
 Name and Identity (Text) Displayed at Start Up
 Distinctive Ringing Based on Calling and Called Number
 Ten User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com
 Speed Dialing
 Configurable Dial/Numbering Plan Support - per Line
 Intercom **
 Group Paging **
 DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy
 Syslog, Debug, Report Generation, and Event Logging
 Secure Call Encrypted Voice Communication Support
 Built-in Web Server for Administration and Configuration with Multiple Security Levels
 Automated Provisioning, Multiple Methods. Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)
 Optionally Require Admin Password to Reset Unit to factory Defaults
** Feature requires support by call server

Documenti

Nessun documento

Links

Link al sito del produttore