Descrierea produsului
Telefon IP , Da , Da , Da , Da , 128 x 64 monochrome LCD graphical display with backlight , SPCP with the Cisco Unified Communications 500 SeriesSIP proxy redundancy: dynamic via DNS SRV, A recordsReregistration with primary SIP proxy serverSIP support in NAT networks (including STUN)SIPFrag (RFC 3420) , Report generation and event loggingStatistics transmitted in BYE messageSyslog and debug server records: configurable per line , Adjustable audio frames per packetDual-tone multifrequency (DTMF), in-band and out-of-band (RFC 2833) (SIP INFO)Flexible dial plan support with interdigit timersIP address/URI dialing supportCall progress tone generation , 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) , Telephone keypad configuration via display menu/navigationAutomated provisioning and upgrade via HTTPS, HTTP, TFTPAsynchronous notification of upgrade availability via NOTIFYNonintrusive in-service upgrades , Jitter buffer: adaptive Frame loss concealmentComfort Noise Generation (CNG)Voice activity detection (VAD) with silence suppressionAttenuation/gain adjustmentsVMWI - Voicemail Waiting Indicator, via NOTIFY, SUBSCRIBE , 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) , Caller ID support (name and number)Third-party call control (RFC 3725)Integrated web server provides web-based administration and configuration , 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) , Secure (encrypted) calling via SRTPCodec name assignmentVoice algorithms:G. 711;G. 726;G. 729 A;G. 722Dynamic payload support , Call waitingCaller ID name and numberOutbound caller ID blockingCall transfer: attended and blindThree-way call conferencing with local mixingMultiparty conferencing via external conference bridge , Highly secure call encrypted voice communications supportBuilt-in web server for administration and configuration with multiple security levels , Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs (STUN) supportDNS SRV and multiple A records for proxy lookup and proxy redundancySyslog, debug, report generation, and event logging , 12 voice lines12 independent SIP RegistrationsLine status: active line indication, with name and numberMenu-driven user interfaceShared line appearanceSpeakerphoneCall holdMusic on hold , 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 , Date and time with support for intelligent daylight savingsCall start time stored in call logsCall timerName and identity (text) displayed at startupDistinctive ringing based on calling and called number , Automatic redial of last calling and last called numbersOn-hook dialingCall pickup: selective and groupCall park and unparkCall swapCall back on busyCall blocking: anonymous and selective , Call forwarding: unconditional, no answer, on busyHot line and warm line automatic callingCall logs (60 entries each): made, answered, and missed callsRedial from call logsPersonal directory with auto-dial (100 entries) , Multiple ring tonesCalled number with directory name matchingAbility to call number using name: directory matching or via caller IDSubsequent incoming calls show calling name and number , 10 user-downloadable ring tonesSpeed dialing, eight entriesConfigurable dial/numbering plan supportIntercomGroup paging , Do not disturbDigits dialed with number auto-completionAnonymous caller blockingUniform Resource Identifier (URI) (IP) dialing support (vanity numbers)On-hook default audio configuration (speakerphone and headset) , 214 x 212 x 44 mm , 0.9 kg , Headset jack: 2.5 mmLED test functionTwo Ethernet ports with integrated Ethernet switch: 10/100BASE-T RJ-45802.3af-compliant PoEOptional 5 VDC universal (100-240V) switching , 4-way rocking directional knob for menu navigationVoicemail message waiting indicator (VMWI) lightVoicemail message retrieval buttonDedicated hold buttonSettings button for access to feature, setup, and configuration menus , Pixel-based display: 128 x 64 monochrome LCD graphical display with backlightDedicated illuminated buttons for:Audio mute on/off;Headset on/off;Speakerphone on/off , Volume control rocking up/down knob controls handset, headset, speaker, ringerStandard 12-button dialing padHigh-quality handset and cradleBuilt-in high-quality microphone and speaker - spa509g
Denumiri similare la Telefon VOIP Cisco SPA509G: SPA 509 G, SPA 509G, SPA509 G

Prețurile și informațiile de pe paginile noastre sunt furnizate de magazinele partenere și au caracter informativ, unele erori pot apărea. Imaginile produselor au caracter informativ, uneori pot include niște accesorii care nu sunt mereu incluse în pachetul de baza. Informațiile aferente produsului (imagine, descriere, preț) se pot schimba fără notificare prealabilă. nu își asumă responsabilitate pentru eventualele greșeli.