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De la Bocris poti cumpara online Telefon VoIP Cisco SPA504G.
Tip: | Telefon IP |
Caller ID: | Da |
Redial: | Da |
Restrictionare apel: | Da |
Data/ora: | Da |
LCD grafic: | 128 x 64 monochrome LCD graphical display with backlight |
Caracteristici speciale: | 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, 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, 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),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), 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), Caller ID support (name and number) Third-party call control (RFC 3725) 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, 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) |
Functii: | Four voice lines 4 SIP registration Line status: active line indication, with name and number Menu-driven user interface Shared line appearance Speakerphone Call hold Music on hold, 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), 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, 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, 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, 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, Highly secure call encrypted voice communications support Built-in web server for administration and configuration with multiple security levels, 10 user-downloadable ring tones Speed dialing, eight entries Configurable dial/numbering plan support Intercom Group paging, 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, 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, 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) |
Dimensiuni baza: | 214 x 212 x 44 mm |
Greutate baza: | 0.9 kg |
Altele: | 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 |