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Cisco IP Phone <br>SPA 509 data sheet Cisco IP Phone
SPA 509 data sheet
Cisco IP Phone <br>SPA 509 User Guide Cisco IP Phone
SPA 509 User Guide
Cisco IP Phone <br>SPA 509 Admin. Guide Cisco IP Phone
SPA 509 Admin. Guide
Quick Start Guide Quick Start Guide
Q & A Cisco IP SPA 509 Q & A Cisco IP SPA 509
Cisco Security Overview Cisco Security Overview
Cisco Unified IP Phone<br> Portfolio - 2013 Cisco Unified IP Phone
Portfolio - 2013
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Cisco IP Phone SPA 509G
Cisco IP Phone SPA 509G big picture
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Product code:
Cisco IP Phone SPA 509G
TVA inclus
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The Cisco Unified IP Phone 500 Series consists of affordable, entry-level IP phones designed for use with the Cisco Unified Communications 500 Series for Small Business. These phones:

• Are an excellent choice for businesses making the transition from traditional digital or hybrid key systems and private branch exchanges (PBXs) to a solution designed for the Cisco Unified Communications 500 Series

• Can be used in cubicles, retail settings, classrooms, break rooms, lobbies, and manufacturing areas

• Are optimized to replace traditional voice systems, or to layer a next-generation voice solution into a Category 5 wired environment

• The Cisco Unified IP Phone 500 Series can also be used with unified communications productivity-enhancing features such as directory search and VoiceView Express.

Full-Featured IP Phones

The Cisco Unified IP Phone portfolio provides:

• IP phones with color LCD displays and dynamic soft keys for call features and functions

• Power efficiency features such as LCD shutoff

• Full-featured phones that support shared line appearance, call park, paging, and intercom

• Security capabilities that allow password-protected configuration changes through the phone



● For business or home office use

● Dependable, Affordable and Feature Rich

• Full-featured 12-line business-class IP phone supporting Power over Ethernet (PoE)

● Monochrome backlit display for ease of use, aesthetics, and on-screen applications

• Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)

• Easy installation and highly secure remote provisioning, as well as menu-based and web-based configuration

• Wideband audio for unsurpassed voice clarity and enhanced speaker quality

• Paper label insert for buttons to indicate speed dials, shared lines, or extensions

• Supports up to two Cisco® SPA500S Expansion Module, adding an additional 64 buttons

• Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco Unified Communications 500 Series for Small Business

Comprehensive Interoperability and SIP-Based Feature Set - Part of the Cisco Small Business Pro Series, the SIP- based Cisco SPA509G Basic 12-Lines IP Phone has been tested to ensure comprehensive interoperability with 2 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 SPA509G 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 (across local and geographically dispersed locations) are just some of the many advantages of the Cisco SPA509G.

The Cisco SPA509G 12-Lines IP phone also supports productivity-enhancing features such as VoiceView Express and Cisco XML applications when used with Cisco Unified Communications 500 Series in SPCP mode.

Carrier-Grade Security, Provisioning, and Management - The Cisco SPA509G 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.

Equipment typeTelefon IP (compatibil Comunicaţii Unificate)
Hardware features● 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
● Dedicated keys for redial, cancel, conference, and transfer
● Standard 12-button dialing pad
● High-quality handset and cradle
● Built-in high-quality microphone and speaker
● Headset jack: 2.5 mm
● 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)
Performed functions● 12 voice lines
● 12 independent SIP Registrations*
● Line status: active line indication
● User interface driven by Interactive Voice Response (IVR)
● Shared line appearance**
● Speakerphone
● Call hold
● Music on hold**
● Call waiting
● 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
● 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 with selectable ring tone per line
● Date and time with support for intelligent daylight savings
● Call start time stored in call logs
● 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

* Feature supported only in SIP mode
** Feature requires support by call server
Data networking● 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)
Voice gateway● 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 μ-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, SUBSCRIBE
● Caller ID support (name and number)
● Third-party call control (RFC 3725)
Administration & 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
Indicator lights/LED● Speakerphone on/off button with LED;
● Headset on/off button with LED;
● Mute button with LED;
● Message waiting indicator LED;
Quality of Service (QoS) OptionsSuportă standardele DSCP (Differentiated Services Code Point) și 802.1Q / p. pentru LAN-uri virtuale şi priorităţi de servicii
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 Advanced Encryption Standard (AES) encryption
● SIP over Transport Layer Security (TLS)
● Secure Real-Time Transport Protocol (SRTP)
Language SupportSuport lingvistic asigurat pentru mai mult de 30 de limbi, în funcție de versiunea utilizată a programului Cisco Unified Communications Manager
Configuration OptionsAlocarea adresei IP poate fi efectuată static sau automat prin DHCP (Dynamic Host Control Protocol)
Standard Interfaces & Ports• 1 Conector RJ-9 - 4-fire, pentru conectarea microfonului şi difuzorului din receptor (handset)
• 2 Conectori RJ-45 - 8 fire, 10/100BaseT Ethernet ports (IEEE 802.3)
• 1 Conector mini jack pentru setul de căști;
• 1 Conector pentru alimentare locala 5 Volti c.c.
Utilities included• Cisco Unified Communications Manager versiunile 4.1(3)sr5b, 4.2(3)sr2b, 4.3(1), 5.1.1 (b), 5.1 (2), 6,0(1) sau mai evoluat (7.x, 8.x)
• Cisco Unified Communications Express ver. 4.1
Supported protocolsCisco Call Control:
• Skinny Client Control Protocol (SCCP)
• Session Initiation Protocol (SIP)
Included licensesNotă: Oricare tip de telefon Cisco Unified IP necesită achiziționarea unei licențe de utilizare pentru tehnologia telefonului, indiferent de tipul protocolul de apel utilizat.
SafetyRegulatory Compliance Products with CE marking indicate compliance with the Low Voltage Directive 2006/95/EC and EMC Directive 2004/108/EC.
ImmunityUL 60950-1, CAN/CSA-C22.2 No. 60950-1, EN 60950-1, IEC 60950-1, AS/NZS 60950-1
Electromagnetic compatibility (EMC)47CFR FCC Part 15B, Industry Canada ICES-003, EN 55022, EN 55024, EN 61000-3-2, EN 61000-3-3, CISPR 22, CISPR 24, AS/NZS CISPR22.
TelecommunicationsFCC Part 68 HAC, CS 03 HAC, AS/ACIF S004, AS/ACIF S040, NZ PTC 220
Power supply typeOpțional
- Adaptor extern în comutație model PA100-EU
Input Voltage100 ~ 240 Vc.a.
AC frequencyminim 47Hz ~ maxim 63Hz
Input current250 mA
Output Voltage5 Vc.c.
Max. output current2 A
Operating temperature0 ~ +40 °C
Storage temperature-20 ~ +70 °C
Operating humidity10 ~ 95% (fără condensare)
Equipment dimensions (w x d x h)214 x 212 x 44 mm
Weight0,9 Kg
ColorCarcasă din plastic (ABS) texturat, culoare gri închis + rama ecranului argintie

● Cisco SPA509G Basic 12-Lines IP phone, handset, and stand
● Handset cord
● RJ-45 Ethernet cable
● CD with:
- Quick-start installation and configuration guide
- User guide
- Administration guide


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Cisco IP Phone SPA 509G :: Cisco VoIP Phones :: CISCO (SPA509G)