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Cisco SPA508G 8-Line IP Phone

Cisco SPA508G 8-Line IP Phone - Hong Kong
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Key Note

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

• Monochrome backlit display

• Wideband audio 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® SP500S Expansion Module, adding up to 64 additional buttons*

• Supports both SIP and SPCP

Comprehensive Interoperability and SIP-Based Feature Set

Part of the Cisco Small Business Pro Series, the SIP-based Cisco SPA508G 8-Line IP Phone 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 SPA508G 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 SPA508G.
The Cisco SPA508G 8-Line IP Phone 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.

Carrier-Grade Security, Provisioning, and Management

The Cisco SPA508G 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.
• SIP version 2 (RFC 3261, 3262, 3263, 326
• 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)

Call control and audio features

• Eight voice lines

• Eight 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 an external conference bridge

• Automatic redial of last calling and last called numbers

• On-hook dialling

• 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

• Hotline 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

• 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, 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

 

• 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)
Security

• 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)

Provisioning, administration

• 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
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 the 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 ports with integrated Ethernet switch: 10/100BASE-T RJ-45

• 802.3af-compliant PoE

• Optional 5 VDC universal (100-240V) switching; the power supply is ordered separately (Cisco PA100)

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 Features
Operating temperature: 32°F to 104°F (0° to 40°C)
Relative humidity:5% to 95% noncondensing, operating and nonoperating
Storage temperature:-13° to 176°F (-25° to 80°C)
Dimensions (H x W x D):• 214 x 212 x 44 mm
Weight: 0.9Kg

 


 

Matrix Technology (HK) Ltd 成立於2009年,香港為總部,主要為粵港澳大灣區提供 IP電話通訊方案,IPVPN 網絡方案,保安系統方案 (香港保安公司牌照 – 第三類別 :1702) 及 視像會議方案 的 通訊科技方案公司。

多年來, 我們將IP電話的應用方案引進到一些中港企業,當中包括:旅行社,商務中心,地產公司,零售商,中港跨境Call Center,學校及一些公共機構當中。我們憑著專業的技術 與 誠懇的服務態度 贏得各界的肯定與信賴。為了提升客戶服務質素,我們在深圳成立了 客戶服務及技術支援中心。除此之外,我們亦與世界各地的VOIP系統顧問公司成為合作夥伴。從而為客戶提供國際性的IP電話系統規劃。為客戶節省了不少時間和成本,並且大大提升了企業營運效率。

Matrix 將在IT,IP通訊 及 保安產品的領域上,承諾持續創新,並且繼續為不同的行業開發合適的VOIP應用方案,為企業創造貼身,可靠 及 高效率 的營商通訊平台為目標。

IPPBX Solution, Telephone system Solution provider in Hong Kong

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Security Company License (TYPE III) | 保安公司牌照 (第三類) : 1702

sgsia 保安局保安及護衞業管理委員會香港持牌保安公司

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Matrix is IP Phone Solution, IP VPN Solution, Video Conference solution & Security Solution Consultancy Company. We based in Hong Kong and Shenzhen,China.

Matrix是一家為香港企業提供 IP電話通訊方案, IP VPN網絡方案, 視像會議方案 及 保安系統 的方案顧問公司