| Telefontype | VoIP-telefon |
| VoIP-protokoll | SPCP, SIP, SIP v2 |
- Three voice lines
- 128 x 64 monochrome graphical liquid crystal display
- Active line indication, name and number
- Menu-driven user interface
- Shared line appearance
- Music on hold
- Caller ID name and number
- Outbound caller ID blocking
- Three-way call conferencing with local mixing
- Multiparty conferencing via external conference bridge
- Automatic redial of last calling and last called numbers
- Call park and unpark
- Call back on busy
- Hot line and warm line automatic calling
- Redial from call logs
- Personal directory with auto-dial
- Digits dialed with number auto-completion
- Anonymous caller blocking
- Support for Uniform Resource Identifier (URI) (IP) dialing
- On-hook default audio configuration
- Called number with directory name matching
- Ability to call number using name: directory matching or via caller ID
- Subsequent incoming calls show calling name and number
- Date and time with support for intelligent daylight savings
- Call duration and start time stored in call logs
- Name and identity displayed at startup
- Distinctive ringing based on calling and called number
- Configurable dial/numbering plan support
- Network Address Translation (NAT) traversal, including Serial Tunnel (STUN) support
- Support for highly secure encrypted voice communications
- Built-in web server for administration and configuration with multiple security levels
- Automated remote provisioning, multiple methods; up to 256-bit encryption
- Dedicated illuminated buttons
- Four-way rocking directional knob for menu navigation
- Voicemail message waiting indicator light
- Voicemail message retrieval button
- Settings button for access to feature, setup, and configuration menus
- Standard 12-button dialing pad
- High-quality handset and cradle
- Built-in high-quality microphone and speaker
- Two Ethernet LAN ports with integrated Ethernet switch
- Password-protected system, preset to factory defaults
- Password-protected access to administrator and user-level features
- HTTPS with factory-installed client certificate
- Up to 256-bit Advanced Encryption Standard (AES) encryption
Based on SIP, the Cisco SPA303 3-line IP phone with 2-port switch helps ensure comprehensive interoperability with equipment from voice over IP (VoIP) infrastructure leaders, enabling service providers to quickly roll out competitive, feature-rich services to their customers. With hundreds of features and configurable service parameters, the Cisco SPA303 addresses the requirements of traditional business users while building on the advantages of IP telephony. Features such as easy station moves and shared line appearances are just some of the many advantages of the SPA303.
The Cisco SPA303 uses standard encryption protocols to perform highly secure remote provisioning and unobtrusive in-service software upgrades. 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 time and expense of managing, preloading, and reconfiguring customer premises equipment.