Cisco SPA512G Dubai

Cisco SPA512G

This phone is compatible with SIP and SPCP protocols, with support for a wide range of call features and functions. Expand the Cisco SPA512G with attendant consoles, or connect a compatible Plantronics headset with EHS. This Cisco SPA500 series phone also includes a backlit LCD and web-based management for easily accessing and administrating the phone.
  • Full-featured 1-line business-class IP phone supporting Power over Ethernet (PoE)
  • Connects directly to an Internet telephone service provider or to an IP PBX
  • Dual Gigabit Ethernet switched ports, speakerphone, caller ID, call hold, conferencing, and more
  • Easy installation and secure remote provisioning, as well as menu-based and web-based configuration
  • Supports up to two Cisco SPA500S Expansion Modules, adding up to 64 additional buttons
Category:

Description

Cisco SPA512G Dubai

The Cisco SPA512G Dubai 1 Line IP Phones with 2-Port Gigabit Ethernet Switch provides the advanced voice and data communications features small businesses need to stay productive and responsive to customers. Part of the Cisco Small Business product portfolio, these stylish, affordable phones offer a rich array of functions, including simple station moves, shared line appearance across dispersed locations, and intelligent call handling and conferencing support. Based on Session Initiation Protocol (SIP), the Cisco SPA512G Dubai ,one line IP phone is fully interoperable with leading voice over IP (VoIP) equipment, for fast deployment and high availability. Highly secure remote provisioning streamlines installation, configuration, and management.

 

Cisco SPA512G Specification

Data networking:
  • MAC address (IEEE 802.3)
  • IPv4 – Internet Protocol v4 (RFC 791)
  • ARP – Address Resolution Protocol
  • DNS – A record (RFC 1706), SRV record (RFC 2782)
  • DHCP Client – Dynamic Host Configuration Protocol (RFC 2131)
  • ICMP – Internet Control Message Protocol (RFC 792)
  • TCP – Transmission Control Protocol (RFC793)
  • UDP – User Datagram Protocol (RFC 768)
  • RTP – Real Time Protocol (RFC 1889) (RFC 1890)
  • RTCP – Real Time Control Protocol (RFC 1889)
  • DiffServ – Differentiated Services (RFC 2475)
  • ToS – Type of Service(RFC 791, 1349)
  • VLAN tagging 802.1p/Q – Layer 2 quality of service (QoS)
  • SNTP – Simple Network Time Protocol (RFC 2030)
Voice gateway:
  • SIP v2 – Session Initiation Protocol version 2 (RFC 3261, 3262, 3263, 3264)
  • SPCP – Smart Phone Control Protocol with UC500
  • 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
  • DTMF – Dual-tone multifrequency, 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
  • VAD – Voice activity detection with silence suppression
  • Attenuation/gain adjustments
  • MWI – Message waiting indicator tones
  • VMWI – Voicemail waiting indicator, via NOTIFY, SUBSCRIBE
  • Caller ID support (name and number)
  • Third-party call control (RFC 3725)
Provisioning, 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
  • TR-69, TR-104, and TR-111 Provisioning
  • Asynchronous notification of upgrade availability via NOTIFY
  • Nonintrusive in-service upgrades
  • Report generation and event logging
  • Statistics transmitted in BYE message
  • Debug and syslog server records: configurable per line
Power supply :
  • Power supply is optional and is purchased separately
  • Models: PA100-NA, PA100-EU, PA100-UK, PA100-AU
  • Switching type (100-240V) automatic
  • DC input voltage: +5 VDC at 2.0A maximum
  • Power adapter: 100-240V50-60 Hz (26-34 VA) AC input
Indicator lights/LED :
  • Speakerphone on/off button with LED
  • Headset on/off button with LED
  • Mute button with LED
  • Message waiting indicator LED
  • Voicemail message retrieval button
  • Hold button

Cisco SPA512G Features

  • One voice line with two call appearances
  • Backlit pixel-based display: 128 x 64 monochrome graphical liquid crystal display (LCD)
  • Line status: active line indication, 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 (100 entries each): made, answered, and missed calls
  • Redial from call logs
  • Personal directory with auto-dial (150 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 duration and start time stored in call logs
  • Call timer
  • Name and identity (text) displayed at startup
  • Distinctive ringing based on calling and called number
  • 12 user-customizable ring tones
  • Speed dialing, eight entries
  • Configurable dial/numbering plan support
  • Intercom
  • Group paging
  • NAT Traversal, including STUN support
  • DNS SRV and multiple A records for proxy lookup and proxy redundancy
  • Advanced Port Mirroring between PC Port and SW Port
  • Debug, syslog, report generation, and event logging
  • Secure call encrypted voice communication support
  • Built-in web server for administration and configuration with multiple security levels
  • Automated remote and secure provisioning via TFTP, HTTP or HTTPS
  • Option to require administrator password to reset unit to factory defaults