Cisco SPA501G 8-Line IP Phone (with POE)

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Cisco SPA501G 8-Line IP Phone (with POE)

產品資料產品規格

Highlights

‧ Full-featured 8-line business-class IP phone supporting Power over Ethernet (PoE)

‧ 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 SPA501G Basic 8-Line IP Phone (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 SPA501G 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 SPA501G.

Carrier-Grade Security, Provisioning, and Management

The Cisco SPA501G 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.

Figure 1. Cisco SPA501G Basic 8-Line IP Phone

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**

‧ 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

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)

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Specifications
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 ‧ 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)	8.42 x 8.35. x 1.73 in. (214 x 212 x 44 mm)
Unit weight	2.00 lb ( 0.9 kg)
Operating temperature	32º ~ 104ºF (0º ~ 40ºC)
Storage temperature	-4º ~ 158ºF (-20º ~ 70ºC)
Operating humidity	5% to 95% noncondensing
Storage humidity	5% to 95% noncondensing