<p>ZX20 series IP phone system is SIP based and optimized for the small and medium business in daily communications. ZX20 series is able to accept 30 user registrations, and easy to manage a fully voice over IP system with the convenient and cost advantages.<br />Based on embedded technology, ZX20 series provides a solid, uniform platform for voice as well as data network communications. It offers a seamlessly integrated solution for the up-to-date telecommunication needs. Being more flexible, ZX20 series integrates 2 analog ports to become a feature-rich PBX system.</p>
<p>ZX20 series integrates NAT functions to make it perfect for small business usage. Besides traditional PBX system functions, it provides many advanced functions including voice mail to email, web management and etc. Designed to run on a variety of VoIP applications, the IP PBX provide IP-based communications, voice conferencing, support paging/intercom, call recording and BLF (Busy Lamp Field) functions. It also supports call detailed record (CDR), centralized Auto-Attendant (AA), and Interactive Voice Responses (IVR). ZX20 series utilizes standard PSTN lines via the interfaces of gateway to support seamless communications among local calls, SIP-based endpoints including low cost long distance call service, telephone number portability and one network for both voice and data.</p>
<h3><strong>Applications</strong></h3>
<ul class="checkbox">
<li>SOHO</li>
<li>Small Businesses</li>
<li>Medium Businesses</li>
<li>Multiple Site</li>
</ul>
<h3><strong><img src="https://www.vensys.pl/media/catalog/product/z/x/zx20app.jpg" alt="" width="100%" /></strong></h3>
<p><strong><img src="https://www.vensys.pl/media/catalog/product/2/0/20110822100624608.jpg" alt="" width="100%" /></strong></p>
<h3><strong>Technical Specification</strong></h3>
<h4>PBX Features</h4>
<ul>
<li>BLF (Busy Lamp Field)</li>
<li>DID (Direct Inward Dialing Number)</li>
<li>Conference Room(3)</li>
<li>Automated Attendant</li>
<li>IVR (Interactive Voice Responses)</li>
<li>DISA (Direct Inward System Access)</li>
<li>CDR (Call Detailed Record)</li>
<li>Voicemail</li>
<li>Feature Codes</li>
<li>SIP Trunk (10)</li>
<li>Blacklist</li>
<li>Phone Book</li>
<li>IP Phone Provisioning</li>
<li>Multiple Language (Chinese / English /Portuguese)</li>
<li>Support Skype for SIP</li>
<li>T.38 FAX(Pass-through)</li>
<li>Trouble Shooting (Ping, Traceroute)</li>
</ul>
<h4>System Capacity</h4>
<ul>
<li>30 IP Phone Users</li>
<li>256 MB Onboard Flash</li>
<li>Recording(GSM): 2100 min</li>
<li>Voicemail(GSM): 2000 min</li>
</ul>
<h4>Call Capacity</h4>
<ul>
<li>Caller ID</li>
<li>Video Calls</li>
<li>Call Paging and Intercom</li>
<li>Call Forward</li>
<li>Call Pickup</li>
<li>Call Park</li>
<li>Call Group</li>
<li>Call Routing</li>
<li>Music on Hold / Transfer</li>
<li>Call Transfer</li>
<li>Call Hold</li>
<li>Call Waiting</li>
<li>Call Queue</li>
<li>Call Recording</li>
<li>Three-Way Conference</li>
</ul>
<h4>Codec & Protocol</h4>
<ul>
<li>Audio Codec:</li>
<li>G.711(a-law, u-law),G.729,G.726,</li>
<li>GSM,Speex</li>
<li>Video Codec Pass-through:</li>
<li>H.261,H.263,H.263+,H.264</li>
<li>Protocol: SIP (RFC3261) , IAX2</li>
<li>DTMF: RFC2833, SIP INFO, In-band</li>
</ul>
<h4>Network Features</h4>
<ul>
<li>DHCP Server</li>
<li>IP Assignment (DHCP / Static)</li>
<li>VPN Client (Support N2N / L2TP)</li>
<li>Network Address Translation (NAT)</li>
<li>DDNS Client (Support Dyndns.org /</li>
<li>No-ip.com)</li>
<li>Support PoE</li>
</ul>
<h4>Hardware Specifications</h4>
<ul class="checkbox">
<li><span style="font-size: 12px;">CPU: 400MHz Blackfin 533 Chip</span></li>
<li><span style="font-size: 12px;">256M On-board NAND Flash</span></li>
<li><span style="font-size: 12px;">64M SDRAM</span></li>
<li><span style="font-size: 12px;">1 WAN Port</span></li>
<li><span style="font-size: 12px;">1 Power Adapter Port</span></li>
<li><span style="font-size: 12px;">1 Reset Button</span></li>
<li><span style="font-size: 12px;">2 Analog Ports</span></li>
<li><span style="font-size: 12px;">LED indication</span></li>
<li><span style="font-size: 12px;">Power Supply:</span></li>
<li><span style="font-size: 12px;">Input: 100 V ~ 240 VAC</span></li>
<li><span style="font-size: 12px;">Output: DC 12 VDC, 2.0 A</span></li>
</ul>
<div>
<p><strong>System</strong></p>
<ul class="checkbox">
<li>Open Source uClinux</li>
<li>Asterisk 1.4.x</li>
</ul>
</div>
<h4>Environmental Specifications</h4>
<ul>
<li>Temperature: -10°C-45°C</li>
<li>Storage temperature:-30°C-65°C</li>
<li>Humidity: 10-80% no dew</li>
</ul>
1Pytania1Odpowiedzi
Pytania
Q
Witam, Mam pytanie ile lini wewnętrznych ta centralka może obsłużyć?Interesuje mnie max. fizyczna liczba portów RJ11 telefonów analogowych i RJ45 telefonów SIP. I oczywiście jak dokonać połączenia tych telefonów do centrali? Z tego co widzę na zdjęciu centralka ma 2xRJ11 i 1xRJ45.
by K_Chrapek on 9 wrz 2014
A
Witam! Wewnętrznych analogowych FXS centrala nie ma, zęwnętrznych FXO ma 2 szt. Wewnętrznych VoIP teoretycznie nie ma ograniczeń, ale producent rekomenduje nie więcej niż 30 szt i 10 jednoczesnych połączeń. VoIP telefony podłączają się przez Ethernet/LAN. Centrala ma RJ-45 gniazdo dla podłączenia switcha/huba. Mamy również VoIP telefony po atrakcyjnej cenie.
Zaloguj się
Stwórz nowe konto