Description

Cisco SPA509G IP Phone Nairobi

The Cisco SPA509G IP Phone in Nairobi is a 12-line phone verified for broad interoperability with foremost producers of VoIP equipment, the SPA509G IP Phone handles workplace integration the way you do; rapidly, competitively, and with feature-rich services.

Traditional and forward thinking at the same time, The SPA509G IP Phone utilizes VoIP’s benefit without compromising its core businesses functionality. The Cisco SPA509G Kenya IP Phone allows easy station moves, productivity improving features when used with Cisco Unified Communications.

The Cisco SPA509G IP Phone works well with Asterisk, Switchvox, Trixbox, 3CX and other SIP based IP PBX platforms. The Cisco SPA509G Kenya supports the standard G.729a codec, as well as G.722 HD Audio and some other voice compression codecs. The Cisco SPA509G IP Phone features a web UI for local provisioning and can also be provisioned remotely via TFTP, HTTP or HTTPS.

Other features of the SPA509G IP Phone include remote in-service firmware upgrade, 256 Bit AES encryption of configuration files, XML configuration files, and asynchronous notification of Upgrade Availability via SIP Notify.

Cisco SPA509G IP Phone Features:

12-Line Phone

Dual switched Ethernet ports

IEEE 802.3af PoE (Power Over Ethernet)
Supports SIP or SPCP protocol
729 and G.722 Wideband Audio Support
Menu driven user interface
Web-based GUI, remote provisioning
IP dialing support
Hi-Res LCD Display
5mm stereo headset connector
AUX port (SPA932 or SPA500S attendant console is supported)
High-quality audio handset
Full-duplex hands-free speakerphone
Hearing aid compatibility (HAC)
Support calls between TTY’s
Works well with GN Netcom, Jabra and Plantronics wired headsets
Optional PA100 Power Adapter and MB100 Wall Mount Adapter

HD Audio
- The Cisco SPA509G IP Phone boasts of a built-in full-duplex speakerphone which supports the use of wideband audio codecs to deliver crystal clear sounds, ensuring that both the listener and the speaker can enjoy clear, natural and life-like conversations. For added security, the audible Dual Tone Multi-frequency (DTMF) tones are masked when the speakerphone mode is used.
Easy User Interface
- Cisco SPA509G IP Phone is an easy-to-use gadget which requires no technical expertise of its user. It is equipped with permanently-labeled buttons which are used to access various functions such as Speaker, Headset, Mute, Volume, Phone, History, Contacts, and Voicemail Message, as well as a permanently-labeled 4-way navigation cluster which enables the user to navigate the screen easily as desired. This saves time therefore maximizing efficiency in the office.
Ultra Clear Backlit Graphical Display Screen
- The Cisco SPA509G IP Phone boasts of a Grayscale 128 x 64 – pixel – based monochrome LCD graphical display for an enhanced visual clarity. The display screen also gives the user a quick access to details since it has programmable backlit line keys and interactive soft buttons that guide you through all call features and functions.

Headset Support
- The Cisco SPA509G IP Phone comes with an integrated headset port with HD Voice support which allows you to directly connect your favorite headset with compatible cable or answer your cordless headset away from your desk with use of a compatible EHS cable. This feature enables you to multitask in the office, thus saving time and maximizing productivity. This also eliminates the need for a separate headset amplifier and allows the handset to remain in its cradle, making headset use simpler.
Supports Power over Ethernet (PoE)
- The Cisco SPA509G IP Phone is comes with dual 10/100Mbps Ethernet switch ports which are used to connect the IP phone to a Power over Ethernet.
Wall Mountable Device
- The Cisco SPA509G IP phone offers an easy installation process. It can be mounted on a wall to achieve a cleaner workstation using an included dual – angle stand. The stand is adjustable to two different angles(supports viewing angles of 30 degrees and 45 degrees) to provide optimum display viewing and comfortable use of all buttons and keys. The foot-stand is keyed to match standard wall-jack configurations for wall mounting. Optional wall-mount brackets are also offered.
Carrier-Grade Security, Provisioning, and Management
- The Cisco SPA509G 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 customer premises equipment.
Compatibility
- The Cisco SPA509G Kenya IP Phone supports for the standard SIP2.0 protocol, as well as Cisco’s proprietary SPCP, a derivative of Cisco’s SCCP (Skinny) call control protocol which is used on the Cisco UC500 platform. It works well with Asterisk, Switchvox, Trixbox, 3CX and other SIP based IP PBX platforms.

Cisco SPA 509G IP Phone Technical Specifications

Telephony Features

12 voice lines
Twelve 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 (callers hear line busy tone)
Digits dialled with number auto-completion
Anonymous caller blocking
Uniform Resource Identifier (URI) (IP) dialling 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
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 dialling, eight entries
Configurable dial/numbering plan support
Intercom
Group Paging
Network Address Translation (NAT) Traversal, 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

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
Standard 12-button dialling pad
High-quality handset and cradle
Built-in high-quality microphone and speaker
Headset jack: 2.5 mm
LED test function
Two Ethernet LAN 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)

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)
Cisco Discovery Protocol (CDP)
Link Layer Discovery Protocol (LLDP)

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 dialling 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
VMWIVoicemail waiting indicator, via NOTIFY, SUBSCRIBE
Caller ID support (name and number)
Third-party call control (RFC 3725)