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Cisco SPA301 1-Line IP Phone

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User Manual for Cisco SPA301 1-Line IP Phone

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Cisco SPA301 1-Line IP Phone Contents of the user manual for the Cisco SPA301 1-Line IP Phone
  • Product category: Telephones Cisco
  • Brand: Cisco
  • Description and content of package
  • Technical information and basic settings
  • Frequently asked questions – FAQ
  • Troubleshooting (does not switch on, does not respond, error message, what do I do if...)
  • Authorized service for Cisco Telephones

User manual for the Cisco SPA301 1-Line IP Phone contain basic instructions that need to be followed during installation and operation. Before starting your appliance, the user manual should be read through carefully. Follow all the safety instructions and warnings, and be guided by the given recommendations. User manual is an integral part of any Cisco product, and if it is sold or transferred, they should be handed over jointly with the product. Following the instructions for use is an essential prerequisite for protecting health and property during use, as well as recognition of liability on the part of the manufacturer for possible defects should you make a warranty claim. Download an official Cisco user manual in which you will find instructions on how to install, use, maintain and service your product.

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Product description

The Cisco SPA301 1-Line IP Phone is part of the Cisco Small Business Pro Series. It 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 SPA301 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 SPA301. The Cisco SPA301 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.
    Telephony Features
  • • One voice line
    • Music on hold*
    • Call waiting
    • Outbound caller ID blocking**
    • Call transfer: attended and blind** ( SIP Only )
    • Three-way call conferencing with local mixing
    • Multiparty conferencing via external conference bridge
    • Automatic redial of last calling and last called numbers
    • Call pickup: selective and group*
    • Call swap
    • Call back on busy**
    • Call blocking: anonymous and selective**
    • Call forwarding: unconditional, no answer and on busy**
    • Hot line and warm line automatic calling
    • Digits dialed with number auto-completion
    • Anonymous caller blocking**
    • Support for Uniform Resource Identifier (URI) (IP) dialing (vanity numbers)
    • Multiple ring tones with selectable ring tone per line
    • Call duration and start time stored in call logs in web GUI
    • 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 Serial Tunnel (STUN) support
    • DNS SRV and multiple A records for proxy lookup and proxy redundancy
    • Syslog, debug, report generation, and event logging
    • Support for highly secure encrypted voice communications
    • 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 1
    Please Note:
    *Feature requires support by call server.
    **Feature activated via feature code
    close
    Hardware Features
  • • Voicemail message waiting indicator light
    • Voicemail message retrieval button
    • Volume control
    • Redial Button
    • Flash Button
    • Standard 12-button dialing pad
    • High-quality handset
    • One Ethernet WAN port 10/100BASE-T RJ-45
    • 5 VDC universal (100-240V) switching power adaptor included
    Security Features
  • • Password-protected system, preset to factory defaults
    • 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
UPC: 882658347504 In the Box Cisco SPA301 1-Line IP Phone
  • Handset Cord
  • RJ-45 Ethernet Cable
  • Power Adapter
  • Quick Installation Guide
  • CD
  • 1 Year Limited Hardware Warranty
  • Guides & Manuals Table of Contents




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    Technical parameters

    Cisco SPA 301 Specs

    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 Protocol (RTP) (RFC 1889, 1890)
    • Real Time Control Protocol (RTCP) (RFC 1889)
    • Real Time Control Protocol - Extended Report (RFC 3611)
    • 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 -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)
    Administration and Maintenance • Integrated web server provides web-based administration and configuration
    • Automated provisioning and upgrade via TFTP, HTTP or HTTPS
    • Asynchronous notification of upgrade availability via NOTIFY
    • Nonintrusive in-service upgrades
    • Report generation and event logging
    • Statistics transmitted in BYE message
    • RTCP-XR
    • Syslog and debug server records: configurable per line
    Physical Interfaces • One 10/100BASE-T RJ-45 Ethernet port (IEEE 802.3)
    • Handset: RJ-9 connector
    Power Supply • Switching type (100-240V) automatic
    • DC input voltage: +5 VDC at 1.0A maximum
    Indicators • Message waiting indicator LED
    Operating Temperature 32 ~ 113ºF (0 ~ 40ºC)
    Storage Temperature -13 ~ 185ºF (-20 ~ 70ºC)
    Operating Humidity 5 to 95% non condensing
    Storage Humidity 5 to 95% non condensing
    Dimension (DxWxH) 1.77 x 7.76 x 3.82" (4.5 x 19.4 x 9.7 cm)
    Weight 1.7 lb ( 0.7711 kg)
    Package Weight 2.0 lb
    Box Dimensions (LxWxH) 11.5 x 8.0 x 2.75"

    Rating

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    Rating: 5 of 5

    This product has already been rated by 04 owners. Add your rating and experience with the product.

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