Cisco SPA502G

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Descrierea produsului
Caracteristici tehnice
Telefon IP Cisco SPA502G  Tip:Telefon IPCaller ID:DaRedial:DaRestrictionare apel:DaData/ora:DaLCD grafic:128 x 64 monochrome LCD graphical display with backlightCaracteristici speciale: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),
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,
Report generation and event logging
Statistics transmitted in BYE message
Syslog and debug server records: configurable per line,
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,
Caller ID support (name and number)
Third-party call control (RFC 3725)
Integrated web server provides web-based administration and configuration,
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,
Secure (encrypted) calling via SRTP
Codec name assignment
Voice algorithms:G. 711;G. 726;G. 729 A;G. 722
Dynamic payload support,
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),
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)
SIP version 2 (RFC 3261, 3262, 3263, 3264),
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)Functii:
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,
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),
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,
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,
Call forwarding: unconditional, no answer, on busy
Hot line 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),
Multiple ring tones
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,
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,
Highly secure call encrypted voice communications support
Built-in web server for administration and configuration with multiple security levels,
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,
10 user-downloadable ring tones
Speed dialing, eight entries
Configurable dial/numbering plan support
Intercom
Group paging, One voice line
One SIP registration
Line status: active line indication, with name and number
Menu-driven user interface
Shared line appearance
Speakerphone
Call hold
Music on holdDimensiuni baza:214 x 212 x 44 mmGreutate baza:0.9 kgAltele:
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,
Volume control rocking up/down knob controls handset, headset, speaker, ringer
Standard 12-button dialing pad
High-quality handset and cradle
Built-in high-quality microphone and speaker,
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 feature, setup, and configuration menus, 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
Denumiri similare la Telefon VOIP Cisco SPA502G: SPA 502 G, SPA 502G, SPA502 G
Păreri
Intrebari&Raspunsuri

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