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Cisco SPA509G

Cisco SPA509G
£145.00

Cisco SPA509G

SKU: SPA-509G
    

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)

• Cisco HD Voice for unsurpassed voice clarity and enhanced speaker quality

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

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

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

Part of the Cisco Small Business Pro Series, the SIP-based Cisco SPA509G 12-line IP Phone with Programmable Keys (Figure 1) has been tested to 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 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 SPA509G.
The Cisco SPA509G 12-line IP Phone with Programmable Keys also supports productivity-enhancing features such as VoiceView Express and Cisco XML applications when used with the Cisco Unified Communications 500 Series in SPCP mode.
Please note - in line with the green policy of Yealink this unit does not include a PSU (Power Supply Unit).

• 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 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

• 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, 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.