OvisLink ePhone-1000s SIP VoIP stolní telefon

The OvisLink ePhone-1000S is a standalone SIP based IP phone with easy-to-setup and excellent voice quality. Therefore, you don't have to turn on your computer 24-hours a day just to receive phone calls as with an USB phone. Now, you can enjoy free IP to IP call almost instantly. It is also compatible with most SIP ITSP (Internet Telephone Service Provider) for reduced call rate to regular phone. Voice Assist Setup The special voice assist setup allow you to plug in the IP Phone directly under your router. The IP phone will automatically get a IP from your router, then you can press the IP button and the phone will announce the IP address to you using voice. The voice assist also including other function as last called number, the phone's number, missed calls, gateway IP, subnet mask and more... So you can know the status of the IP phone without ever need to use the web configuration. Simple Setup With both Outbound Proxy and STUN server capability, the ePhone-1000S can easily work under NAT routing environment. Therefore, you can plug in multiple IP phones under your router, then it will work instantly without the need to open any virtual server port in your router. Of course, it will also work with fixed IP and PPPoE connections. For configuration, you can use the IP Phone through phone's keypad or web browser. Furthermore, setup guide with popular free Proxy Service are provided. Comprehensive Phone Features The phone have additional feature such as speed dial, volume control, and many other additional functions. In addition, standard earphone and microphone jack allows user to attach accessories to the phone. Or you can press the speaker phone button and talk hands free. Specification Hardware - Data Memory: 16MB - Procedure Memory: 8MB Flash memory - NAT Pass-through (SIP with STUN) Connectors - 1 x RJ-45 LAN port - Headphone jack - MIC jack Voice - SIP (RFC-3261) - Compression:G.711 A/µ -Law, G.723.1, G.729A/B/AB - Support Silence Suppression, VAD, CNG, Acoustic Echo Cancellation, Jitter Buffering - Echo Cancellation: G.165 16ms - Delayed (End to End):< 100ms - Flow of the average:10-12k bit/s - DTMF tone detection E.164 Dial plan Networking Protocol - PPPoE - DHCP client/Static IP - UDP/TCP - RTP/RTCP - FTP/HTTP/DNS Management Functions - Web browser based management - Phone button configuration Calling Function - Call Hold, Call Mute, Call Status, Caller ID - Headset Connector - Speaker Phone - Networking Status - Phone Book - Volume Adjustment - No-Answer Call Forward - Busy Call Forward, - Always Call Forward - Miss-Call, Dialed-Call, Answered-Call - Last Number Redial - Speedy Dial Display - 32 characters LCD display(2 lines with 16 characters each) Voice Assist buttons - Called - Answered - Miss-Call - Phone - Gateway IP - IP - Subnet Mask - Server IP Other Buttons - Speedy-Dial - Call-Hold - Redial, Mute, Volume Up/Down - Speaker Phone Power - Input AC Range: 100 – 240VAC, 50 – 60Hz - Input DC Range: 5VDC, 1A - Power Consumption: 3W Environmental - Operation Temperature: 0 - 40? (32° - 104°F) - Storage Temperature: -30 - 65? (-22° - 149°F) - Relative Humidity: 10 – 95% Non-Condensing Safety Compliance - CE - FCC