Origin	China
Model Number	IP PBX S800
Brand Name	Uni-Ta

Asterisk Powered IP PBX Applicance for SMBs: To provide a complete turnkey solution and meet our customers’ demands on modern business communication, Uni-Ta proudly introduces the Asterisk embedded IP PBX system Univox S800, which is based on uClinux/Blackfin. By using state-of-the-art embedded technology, the IP PBX S800 provides a solid, uniform platform for voice communications as well as data network communications.

Offering rich IP PBX features such as customizable Auto Attendant & IVR, call park/hold/retrieve/waiting/forward, call queues, hunt/ring group, music on hold, conference bridges, voicemail, plus built-in SIP/IAX server and automated NAT traversal, the IP PBX S800 enhances your communication efficiency and makes your business more productive. The application of flexible dial plan and SIP/IAX trunk or analog trunk will eliminate communication cost to the lowest level.

The IP PBX S800 is equipped with 8 FXO/FXS channels and 2 Ethernet ports (NAT router). Enterprises who are not yet ready to migrate to a complete VoIP solution can connect the IP PBX system to PSTN or traditional PABX system. Integrated with Uni-Ta’s VoIP phones and VoIP gateways, the IP PBX S800 is an ideal solution for small & medium businesses (sub-100 users) and remote branch offices of larger organizations at very affordable cost.

Features & Benefits

- Asterisk powered Embedded IP PBX (open source)

- System: uClinux/Astfin

- Intuitive web-based GUI

- Fully support SIP & IAX2 protocols

- Unlimited SIP/IAX2 user extensions and supports 30 concurrent calls

- Integrated PSTN trunks, analog phone/FAX ports & SIP/IAX trunk options

- Flexible dial plan

- Excellent OSLEC (Open Source Line Echo Cancellation)

- Up to 8 FXO/FXS channels

- Support for a combination of FXO and FXS modules

- Built-in router ideal for small offices

- MMC/SD slot for external storage

- RS-232 port for console configuration

Basic Call Services

- Caller ID

- Call Waiting

- Call Park and Retrieve

- Call Hold and Retrieve

- Call Pickup

- Call Forward (Busy, No Answer, Uncondicontional)

- Call Transfer

- Call Logs

- Auto Attendant & powerful IVR

- Call Queues

- Hunt/Ring Group

- Music on Hold

- Conference Bridges (up to 8 lines per conference)

- Voicemail (up to 2G with expansion storage)

Standards & Voice Codec

- Standards: SIP 2.0 (RFC 3261); IAX2

- Voice codec and supports DSP trans-coding: G.729, G.711 a-law/ u-law, G.726, GSM, Speex

- DTMF Relay: RFC2833, SIP Info

Network & Management

- Connection mode: DHCP, Static, PPPoE

- NAT traversal

- Intuitive Web-based GUI for system configuration

Hardware Specification

- CPU: Blackfin 533 Chip, 400MHz

- Memory: 64MB SDRAM

- Storage:

*Default NAND Flash: 256M

*Storage expansion MMC/SD slot

- Network Interfaces:

* WAN: 1-RJ45 (10/100Base-TX, auto-sensing)

* LAN: 1-RJ45 (10/100Base-TX, auto-sensing)

- RJ11 port X 8

- RS-232 port X 1 for console configuration

- MMC/SD slot X 1 for expansion storage

- USB port X 1

- Power Jack X 1