Issuu on Google+

SIP Connect IP PBX Appliance The next generation in business communication system, an innovative all-in-one voice communication solution developed for small to medium sized business & companies with up to 50 seats per location out-of-the-box. Designed from ground up to support distributed IP communication with advanced application integration and common PBX features cater for both PSTN and ISDN connection.

System Feature Highlights  Integrated PSTN interface card with 4 FXO ports for analog phone lines (Optional 8 FXO ports)  Integrated ISDN interface card with E1/T1 port for digital lines  Immediate provides VoIP connection to your local telco phone lines and to any remote site  Unified messaging with voicemail to email attachment supported  Automated provisioning of IP Phones, video phones, ATA and other endpoints for easy deployment  Remote management capability  Virtual conference room service to expand meeting room requirement  Web-based wizard function for easier configuration with Integrated voice mail to manage your voice message efficiently  Most legacy PBX features supported

Call features • Unlimited extensions, 50 ready-configured extensions • Codec G.711 (μ/A-law) supported • Optional G.729A and G.726 (16k/24k/32k/40k bit/s) separate license required • Enable/Disable NAT Traversal per SIP trunk • Call admission control of call count or bandwidth per SIP trunk • Support SIP OPTIONS keep alive, NAT session keep alive • Configurable RFC 2833 payload type per SIP trunk • FXS/FXO analog trunking • FXO disconnection tone detection • Caller ID detection (service from telco end required) • Trunk hunting • Support SIP Call Hold, Call Waiting • Support SIP phone 5 3-way conference • Support Blind/ Attended Transfer • In-line Call Transfer • Unconditional, Unavailable, Busy Call forward • Call Back on Busy between extensions • Per calling number forward and rejection • Blacklist of number patterns • Multiple setting for call pick-up groups • Call Park and Retrieve • Remote extension registration via Internet • Echo Cancellation (G.168) • Flexible numbering plan • Call privilege grouping

August 2010

 


PBX System Call Features • Unlimited users supported • 50 users-ready extensions with voice mail accounts • 16 concurrent sessions • Codec G.711 (μ/A-law), G.723.1 supported • Optional G.729A separate license required • Support gateway trunk mode per SIP trunk • Enable/Disable NAT Traversal per SIP trunk • Call admission control of call count or bandwidth per SIP trunk • Support SIP OPTIONS keep alive • NAT session keep alive • FXS/FXO analog trunking or E1/T1 digital trunking • Caller ID detection (service from telco required) • Trunk hunting • Support SIP Call Hold, Call Waiting • Support SIP phone 3-way conference • Support Blind/ Attended Transfer • Unconditional, Unavailable, Busy Call forward • Call Back on Busy between extensions • Per calling number forward and rejection • Multiple setting for call pick-up groups • Call Park and Retrieve • Remote extension registration via Internet • Flexible numbering plan • Call privilege grouping Administration System Management • Web-based configuration with session control • User and administrator configuration mode • Automatic expiring the idle sessions • Support firmware upgrade through the Internet • Configuration Wizard for mass extensions and users creation • System event Syslog • Downloadable Call Detail Record (CDR) • Extension registration status • Active call status • NTP synchronization • Real Time Clock setting • DHCP Server • Configurable Time Zone • Firmware Upgrade through web interface

Voice Mail • User Authentication by PIN • Multi-folder Archive • Fast-forward /Rewind/Undelete • E-mail notification and attachment • Personal greeting on unavailability and busy • Record personal greeting through phone • Voicemail Forwarding • Reply call or new call after logged in Voicemail • Support USB 2.0 interface for Voicemail, CDR, and system configuration backup Virtual Conference Room • Up to 8 conference rooms with configurable number and PIN • Up to 8 parties calls among all conference rooms • Music on First Dial-in Party • Hot key to leave the conference • Hot key for administrator to manage the conference

Hardware Specification Hardware Interfaces • One RJ-45 10/100 Base-T Ethernet ports • 4x FXO PSTN interface model or 1x E1/T1 ISDN interface model • Two USB 2.0 ports • One RS-232 serial port • One VGA port • Two PS/2 ports (Keyboard, Mouse) System Dimension • Small Form Factor Casing • 300 x 155 x 38 (mm) System Power Requirement • Power input 100~240V AC, 50~60 Hz • 65W AC adapter / 90W (optional) Environment • Operating temperature 0~50Ԩ • Storage temperature -10~70Ԩ • Humidity (RH) 10~80% non-condensing

Interactive Voice Recognition (IVR) • User configurable IVR • Work time/Holiday setting for different IVR • Configurable greeting prompts • Music on Ring extensions • Forward to Voice Mail on No-answer

  Selection of IP Phones, IP Conference Phone, Headset & Accessories 

  Accede Technology Pte Ltd 10 Ubi Crescent, #04-96/97 UBI Techpark, Lobby E Singapore 408564 Tel: (65) 6593 7711 Fax: (65) 6848 1183 Email: enquiry@accede.com.sg

August 2010

 


SIP Connect Brochure