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Why IP PBX? |
IP PBXs have become the new standard in business communications systems, merging the call routing and processing features of a traditional telephone system with the flexibility and global reach of the Internet.
IP PBX systems, VoIP gateways, conference servers and more. It is used by small businesses, large businesses, call centers, carriers and governments worldwide. |
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SIP Connect IP PBX appliance
SIP Connect is designed on Asterisk platform that turns an appliance into a voice communications server. SIP Connect powers IP PBX system, VoIP gateways, conference and bridging used by small business, medium to large organisation leveraging on the latest technology.
SIP Connect comes with two models to cater for analog PSTN phone lines and digital ISDN lines. Connecting with a wide selection of hardware IP phones and soft-phones, call recording features, inter-active voice recognition and voice messages with Outlook or Outlook Express. |
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| BENEFITS |
SIP Connect IP PBX appliance is the next generation in business phone system.
• The innovative all-in-one voice communication solution developed for small to medium sized business, companies with up to 50 seats ready 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 either PSTN or ISDN connection.
• Connects to your existing IT network infrastructure delivers the telephony needs for your office with no separate wiring required.
• Designed on open standards, you can selected from a wide range of phone sets, from any industry standard SIP hard phones, soft phones and IP conference phone sets and flexibility.
• Offers modern business communication system new height of innovation, quality reliability feature integration user productivity ease of deployment & administration and affordability. |
TYPICAL IP PBX ENVIRONMENT
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| PBX SYSTEM Specification |
Call features
• 50 users and extensions with voice mail account
• 16 concurrent sessions
• Codec G.711 (μ/A-law), G.723.1 (6.3k/5.3k bit/s) supported
• Optional G.729A and G.726 (16k/24k/32k/40k bit/s) separate license required
• 50 DID SIP trunks to extensions (expandable)
• 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
• 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 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
• Configurable Music on Hold
• Support T.38 FAX over IP
• Support T.30, T.38 FAX pass through
• ENUM resolution |
ISDN lines supported
• Direct line to extension (DID to Extension)
• Direct line by called number (DID by Number)
• Direct line by privilege (DID by Privilege)
IVR
• Worktime/Holiday setting for different IVR
• Configurable greeting prompts
• Music on Ringing extensions
• Forward to Voice Mail on No-answer
• Hot key to operator
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 menu
• 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
• Lock/Mute/Join/Drop control for administrator
• Music on First Dial-in Party
• Hot key to leave the conference
• Hot key for administrator to manage the conference |
| ADMINISTRATION |
HARDWARE SPECIFICATION |
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
• Step-by-Step Wizard for adding users, extensions and trunks
• Built in online help in wizard
• Command Line Interface (CLI) for configuration
• System event Syslog
• Downloadable Call Detail Record (CDR)
• Extension registration status
• Active call status
• TFTP server and TFTP repository maintenance
• NTP synchronization
• Real Time Clock setting
• DHCP Server
• Configurable Time Zone
• Firmware Upgrade through web interface and console |
Hardware interfaces
• One RJ-45 10/100 Base-T Ethernet port
• Four FXO PSTN or Four FXO/FXS
interface model or
• One E1/T1 ISDN interface model
• Two USB 2.0 port, One RS-232 serial port
• One VGA port, Two PS/2 ports (Keyboard, Mouse)
System Dimension (SFF Casing)
• 300 x 155 x 38 (mm)
System Power Requirement
• Power input 100~240V AC, 50~60 Hz
• 20 W (max)
Environment
• Operating temperature 0~50℃
• Storage temperature -10~70℃
• Humidity (RH) 10~80% non-condensing |
| For End-User enquiry : |
Call (65) 6593 7711 |
| For Partner enquiry : |
Call (65) 6603 3333 |
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