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Quick Details

  • Place of Origin: Guangdong, China (Mainland)
  • Brand Name: RealTone Technologies,Inc
  • Model Number: RT80-12S/12
  • Type: VoIP Gateway
  • Certifications: FCC CE
  • Colour: Black
  • Product name: IPPBX
  • Voice Codec: G.711 A/ulaw, G.729,iLBC,GSM
  • FXS port: 12
  • FXO port: 12
  • SIP extensions: 60
  • SIP trunk: 80
  • Protocol: SIP
  • Warranty: 12 Months

Packaging & Delivery

Packaging Details: Carton Package, Size: 55*41*37cm;
Delivery Detail: 3days

 

RT80 is an all-in-one IP-PBX for organizations with up to 100 employees in one location or multiple locations, featuring built-in analog phone interfaces, CO line interfaces, SIP trunks and SIP client support. With features such as click-to-call, auto-attendant, on a fly recording, mobile extension, and call back, it helps to reduce operation cost and improves the communication with customers.

 

Applications

- Telephon system for SMB

- Call center

- Telephone system for virtual offices

 

Features:

Attendant
Automated attendant
Interactive voice response (IVR)
Working hour and non-working hour service
Changing automated attendant profile on a fly
Queuing
Operator group (Up to 5 operators)
Operator console

Incoming call
Direct inward dialing (DID)
Direct dialing in (DDI)
Calling line identification (CLI) distribution
Call block
Ring tone pattern selection
Call waiting

Group receiving
Extension group
Group call distribution
Group ringing

Extension receiving
Call forking (Ring simultaneously another pre-selected terminal)
Call forwarding (FWD)
Do not disturb (DND)
Ring pattern selection
Color ring back tone

Call pickup
Directed pickup
Group pickup
Pickup operator’s call only

Making call
Intercom call
Account code entry
Outgoing line preference
Trunk group
Speed dialing
Hot line (Immediate, delay)
Caller ID block
Emergency call

Outside access
Direct inward system access (DISA)
Call back
Remote extension setting on phone

Call restriction
Toll restriction/Call barring
Budget management
Extension lock
Verified code entry

Routing

Automatic route selection

Call transfer
Blind call transfer
Consultation call transfer
Call transfer to CO line

Call hold
Call hold
Call park
Music on hold

Conference
3-way calling

Caller ID
Caller ID
Incoming call log

Message notification
Voicemail message waiting (FSK, polarity reversal)

Recording
Extension recording
On the fly recording

Audible tone
Dial tone
Confirmation tone
Stutter dial tone

CO line (FXO)
Polarity inverse detection
Caller ID detection
Busy tone detection
DTMF outpulsing delay
Volume control

Subscriber line (FXS)
Polarity inverse generation
Caller ID generation (FSK, DTMF)
Ring cadence setting
Ring frequency setting
Volume control

SIP trunk
Registration per line
Registration per system

Tie trunk service
Direct internal dialing
Call forwarding
Call transfer
Outbound trunk sharing

Other SIP connection
Peer-to-peer SIP communication

SIP extension
Outgoing
Incoming
HOLD
Transfer
Busy line field (BLF)

PC assistant
Click-to-call
Contact list
Presence
Short message
Retrieving voice mail and recording file
Call log

Networking
DHCP
DNS/DDNS
PPPoE
NAT traversal
IP address filtering for security

Media stream processing
Codecs: G.711, G.729A, T.38
Echo cancellation
Jitter buffer automatic adjustment
Busy tone detection

Non-telephone terminal
Fax transmit and receive (T.30 and T.38)
POS transmit
PBX connectivity (Analog trunk and analog extension)

System management
Web management interface (Local and remote access)
Firmware upgrade
Log management
Configuration import and export
System status monitoring and statistics

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