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PLANET IPX-2000 is the next generation voice communication platform for the small to medium enterprises. Designed as an open, scalable and highly reliable telephony solution, the IPX-2000 is able to accept 200 extension registrations, and effectively scales from under 100 users to as many as 200 in a standard rack-mountable unit. PLANET IPX-2000 is also designed to operate on a variety of VoIP applications; it provides centralized call control, auto-attendant, voice conferencing, and PSTN access, digital and IP-based communications.

The IPX-2000 integrates up to 8 calls via the IPX-FXO (Foreign eXchange Office, FXO) module to become a feature-rich PBX system that supports seamless communications between the existing PSTN calls, analog, IP phones and SIP-based endpoints.

Moreover, the IPX-2000 also integrates telephony call processing, call control, voice mail, and a wide PBX application programming interface into a highly scalable architecture designed to support both the traditional circuit-based and the Internet telephony service within a distributed enterprise communications network.

With the IPX-2000, standard SIP phones can be easily integrated in your office, plus the auto-config feature, which may be integrated with PLANET IP phone (VIP-254 series), and the analog telephone adapter (VIP-156 / 157 series) to build the VoIP network deployment in minutes.

Allowing distributed IP technology to meet the traditional voice services, with the IPX-2000 proactive management interface in the daily business process, it brings enterprises higher employee productivity and customer satisfaction.

Allowing distributed IP technology to meet traditional voice services, with the IPX-2000 proactive management interface in the daily business process, it brings enterprises higher employee productivity and customer satisfaction.


VIP-156 + VIP-351PT IPX-2000 integration

IPX-2000 intra office voice communication
Key Features

System Highlight

  • 200 max users / extensions with voicemail account
  • 50 max concurrent sessions
  • Highly integrated, embedded system for stability
  • Immediate VoIP connections
  • Analog interface (FXO / FXS) support
  • Flexible dialing plans
  • On-ramp / off-ramp calls for VoIP and PSTN
  • Remote management
  • Wizard for easy configuration
  • Command Line Interface (CLI) for quick-batch configuration
  • Seamless integration with legacy PBX
  • Auto provision and auto firmware upgrade for feature IP phones

Feature Highlight
  • SIP registrar and SIP proxy
  • Multi-language voice prompts for international business
  • Customizable 3-layer IVR
  • ACD (Automatic Call Distribution) to form basic Call Centers
  • Stackable design for scalability and investment protection
  • UMS (Unified Messaging System) support
  • Meet-me conference for virtual meeting room
  • System status and system log
  • Call Detail Record (CDR)
  • Built-in STUN client
  • Function-rich voicemail System
  • CAC (Call Admission Control)
  • Call keep alive


Interface One RJ-45 10/100 base-TX WAN Ethernet port
One RJ-45 10/100 base-TX LAN Ethernet port
Two USB 2.0 ports
One RS-232 serial port
Two expandable PCI interface slots
Standards and protocol
Registration Max. 200 nodes / SIP IP phones
Calls Max. 50 concurrent calls
Call control RFC 3261, RFC 3311, RFC 3515, RFC 3265, RFC 3892, RFC 3361, RFC 3842, RFC 3389, RFC 3489, RFC 3428, RFC 2327, RFC 2833, RFC 2976, RFC 3263, RFC 3264, RFC 3362, RFC 4612
SIP Registrar Static / Dynamic registration, Configurable Expiry Time, MD5 authentication
Handle loose RFC-compliant SIP devices, Resilient message retry mechanism
Cache client registrations
SIP Proxy Stateful proxy server, NAT traversal for clients, Inter-proxy call hand-off
Outbound Proxy behind NAT Device
PBX System
Call features Codec G.711 (μ/A-law), G.723.1 (6.3k/5.3k bit/s), G.729A, and G.726 (16k/24k/32k/40k bit/s) supported
Transcoding channel 0~16, subject to add-on card
In-band / RFC2833 / SIP-INFO DTMF translation
Two expandable slots for telephony interfaces
50 SIP trunks for ITSP account or private trunking shared by extensions
200 DID SIP trunks to extensions
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
Long call audit
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
FXO disconnection tone parameter setting
FXS hot line
FXS warm line
Caller ID detection
Trunk hunting
Digits manipulation during hunting
Life-line priority call
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
32 call pick-up groups
Call Park and Retrieve
Recording on demand with PLANET IP phones
Remote extension registration via Internet
Direct line to extension (DID to Extension)
Direct line by called number (DID by Number)
Direct line by privilege (DID by Privilege)
Echo Cancellation (G.168)
Flexible numbering plan
Call privilege grouping
Configurable Music on Hold
Memo Call for extension
Schedule-based Broadcast
Support T.38 FAX over IP
Support T.30, T.38 FAX pass through
ENUM resolution
NAT Auto NAT discovery and traversal
Built-in STUN client
RTP proxy
RTP port range designation
IVR 50 configurations of 3-layer IVR
Worktime / Holiday setting for different IVR
Configurable greeting prompts
Music on Ringing extensions
Forward to Voice Mail on No-answer
Support 3 languages in IVR tree
Hot key to operator
Voice User Authentication by PIN
Multilingual, 3 languages
Multi-folder Archive
Fast-forward / Rewind / Undelete
MWI notification
VMWI notification
E-mail notification and attachment (Unified messaging)
Personal greeting on unavailability and busy
Record personal greeting through phone
Voicemail Forwarding
Reply call or new call after logged in Voicemail menu
Built-in 40GB hard disk drive for Voicemail
Support USB 2.0 interface for Voicemail, CDR, and system configuration backup
Support NFS remote backup for Voicemail, CDR, and system configuration
Meet-me conference 24 conference rooms with configurable number and PIN
Up to 24 parties 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
Automatic Call Distribution 32 queues with 32 agents among all queues.
32 inbound call among all queues
Configurable waiting length for individual queue
Support five distribution policies including round robin, ring all, least recent, fewest call, and random
Configurable waiting time for each queue
Allow agent remotely log-in
Agent can participate multiple queues
Agent phones also allow extension calls
Stackable Support LAN stacking up to 4 units in the same model
Automatic intra-trunking creation among stacking units
Automatic configuration publishing from Master to Slaves
Automatic load balancing in hosting feature phones3
Internet Sharing
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
Network management DHCP / PPPoE / Static IP on WAN
Support MAC Clone on WAN
Static LAN routing
Firewall on predefined services
Virtual Server for client device
NAT for outbound traffic from LAN
WAN QoS queuing mechanism for VoIP and data traffic
Support TOS setting
DNS forwarder and dynamic DNS
SNMPv2 with standard MIB format
Adaptive WAN bandwidth and DSP channel saving
Operating Temperature Operating temperature 0~50℃
Storage temperature -10~70℃
Humidity (RH) 10~80% non-condensing
Power Requirement 100~240V AC, 50~60 Hz