Cisco SPA-501G 8-Line IP Phone with 2-Port Switch, PoE and Paper Label
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80,56 €
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Telephony Features

Eight voice lines
Four Independent SIP Registrations
Line status: active line indication
User interface driven by Interactive Voice Response (IVR)
Shared line appearance
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
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

Hardware Features
Paper label area
Dedicated illuminated buttons for: – Audio mute on/off – Headset on/off – Speakerphone on/off
Voicemail message waiting indicator (VMWI) light
Voicemail message retrieval button
Dedicated hold button
Settings button for access to IVR menu
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)

Regulatory Compliance
FCC (Part 15, Class B), CE Mark, A-Tick, C-Tick, Telepermit, UL, CB

Security Features
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)
Τεχνικά χαρακτηριστικά
Τεχνικά χαρακτηριστικά

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
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
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
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

Power supply

Power supply is optional and is purchased separately
Models: Cisco PA100-NA, PA100-EU, PA100-UK, PA100-AU
DC output voltage: +5 VDC at 2.0A maximum
Switching power adapter: 100-240V 50-60 Hz AC input

Physical interfaces

Two 10/100BASE-T RJ-45 Ethernet ports (IEEE 802.3)
Handset: RJ-9 connector
Built-in speakerphone and microphone
Headset 2.5mm jack

Indicator lights/LEDs

Speakerphone on/off button with LED
Headset on/off button with LED
Mute button with LED
Message waiting LED

Body dimensions (W x H x D)

214 x 212 x 44 mm

Unit weight

0.9 kg
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