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The SPA509G 12-Line IP Phone with 2-Port Switch PoE and LCD Display, which is part of the Cisco Small Business Pro Series, is a dependable and affordable IP phone for business or home office use. The Cisco IP Phone has been tested to ensure comprehensive interoperability with equipment from voice over IP (VoIP) infrastructure leaders, enabling service providers to quickly roll out services to their customers.
With hundreds of features and configurable service parameters, the Cisco IP Phone caters to the needs of traditional business users with features such as easy station moves and shared line appearances (across local and geographically dispersed locations).
The Cisco IP Phone 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 the customer's premises for equipment.
• 12 voice lines• 12 Independent SIP registrations• Line status: active line indication, with name and number• Menu-driven user interface• Shared line appearance• Speakerphone• Call hold• Music on hold• Call waiting• Caller ID name and number• Outbound caller ID blocking• Call transfer: attended and blind• Three-way call conferencing with local mixing• Multiparty conferencing via external conference bridge• Automatic redial of last calling and last called numbers• On-hook dialing• Call pickup: 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• Digits dialed with number auto-completion• Anonymous caller blocking• Uniform Resource Identifier (URI) (IP) dialing support (vanity numbers)• On-hook default audio configuration (speakerphone and headset)• Multiple ring tones• 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 start time stored in call logs• Call timer• Name and identity (text) displayed at startup• Distinctive ringing based on calling and called number• 10 user-downloadable ring tones• Speed dialing, eight entries• Configurable dial/numbering plan support• Intercom• Group paging• Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs (STUN) support• DNS SRV and multiple A records for proxy lookup and proxy redundancy• Syslog, debug, report generation, and event logging• Highly secure call encrypted voice communications support• Built-in web server for administration and configuration with multiple security levels• Automated remote provisioning, multiple methods; up to 256-bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP])• Option to require administrator password to reset unit to factory defaults
• Pixel-based display: 128 x 64 monochrome LCD graphical display with backlight • Dedicated illuminated buttons for: - Audio mute on/off - Headset on/off - Speakerphone on/off • 4-way rocking directional knob for menu navigation • Voicemail message waiting indicator (VMWI) light • Voicemail message retrieval button • Dedicated hold button • Settings button for access to feature, setup, and configuration menus • Volume control rocking up/down knob controls handset, headset, speaker, ringer • Standard 12-button dialing pad • High-quality handset and cradle • Built-in high-quality microphone and speaker • Headset jack: 2.5 mm • LED test function • Two Ethernet ports with integrated Ethernet switch: 10/100BASE-T RJ-45 • 802.3af-compliant PoE • Optional 5 VDC universal (100-240V) switching; power supply is ordered separately (Cisco PA100)
• 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 Advanced Encryption Standard (AES) encryption • SIP over Transport Layer Security (TLS) • Secure Real-Time Transport Protocol (SRTP)
In the Box
Cisco SPA509G 12-Line IP Phone with 2-Port Switch PoE and LCD Display
RJ-45 Ethernet Cable
1 Year Limited Hardware Warranty
Table of Contents
MAC address (IEEE 802.3) IPv4 (RFC 791) Address Resolution Protocol (ARP) DNS: A record (RFC 1706), SRV record (RFC 2782) Dynamic Host Configuration Protocol (DHCP) client (RFC 2131) Internet Control Message Protocol (ICMP) (RFC 792) TCP (RFC 793) User Datagram Protocol (UDP) (RFC 768) Real-Time Transport Protocol (RTP) (RFC 1889, 1890) Real-Time Control Protocol (RTCP) (RFC 1889) Differentiated Services (DiffServ) (RFC 2475) Type of service (ToS) (RFC 791, 1349) VLAN tagging 802.1p/Q: Layer 2 quality of service (QoS) Simple Network Time Protocol (SNTP) (RFC 2030)
SIP version 2 (RFC 3261, 3262, 3263, 3264) SPCP with the Cisco Unified Communications 500 Series SIP proxy redundancy: dynamic via DNS SRV, A records Reregistration with primary SIP proxy server SIP support in NAT networks (including STUN) SIPFrag (RFC 3420) Secure (encrypted) calling via SRTP Codec name assignment Voice algorithms: G.711 (A-law and -law) G.726 (16/24/32/40 kbps) G.729 A G.722 Dynamic payload support Adjustable audio frames per packet Dual-tone multifrequency (DTMF), in-band and out-of-band (RFC 2833) (SIP INFO) Flexible dial plan support with interdigit timers IP address/URI dialing support Call progress tone generation Jitter buffer: adaptive Frame loss concealment Comfort Noise Generation (CNG) Voice activity detection (VAD) with silence suppression Attenuation/gain adjustments VMWI - Voicemail waiting indicator, via NOTIFY, UBSCRIBE Third-party call control (RFC 3725)
Administration and Maintenance
Integrated web server provides web-based administration and configuration Telephone keypad configuration via display menu/navigation Automated provisioning and upgrade via HTTPS, HTTP, TFTP Asynchronous notification of upgrade availability via NOTIFY Nonintrusive in-service upgrades Report generation and event logging Statistics transmitted in BYE message Syslog and debug server records: configurable per line
Two 10/100BASE-T RJ-45 Ethernet ports (IEEE 802.3) Handset: RJ-9 connector Built-in speakerphone and microphone Headset 2.5 mm jack
Power supply is optional and is purchased separately DC output voltage: +5 VDC at 2.0A maximum Switching power adapter: 100-240V 50-60 Hz AC input
Speakerphone on/off button with LED Headset on/off button with LED Mute button with LED Message waiting LED