PLANET IPX-1800 series IP PBX system are designed and optimized for the SMB, and SOHO daily communications. The IPX-1800 is the next generation voice communication platform for the small to medium enterprise. Designed as an open, scalable, and highly reliable telephony solution, the IPX-1803 is able to accept 30 extension registrations, and effectively scales from under 30 users to as many as 50-user enterprise. Designed to run on a variety of VoIP applications, the IPX-1803 provides centralized call control, auto-attendant, voice conferencing, PSTN access, and IP-based communications.

The IPX-1800 series divided into two models: the IPX-1803 integrates up to 4 telephony interfaces, including 3 FXO (Foreign eXchange Office, FXO), 1 FXS (Foreign eXchange Station, FXS); the FXS interface in IPX-1803 provides lifeline functionality; the IPX-1804 integrates 4 FXO telephony interfaces to become a feature-rich PBX system that supports seamless communications between existing PSTN calls, analog, IP phones and SIP-based endpoints.

The IPX-1803 integrates telephony call processing, call control, voice mail, and a widely PBX application programming interface into a highly scalable architecture designed to support both traditional circuit-based and the Internet telephony service within a distributed enterprise communications network.

With IPX-1803, standard SIP phones can be easily integrated in your office; plus the auto-config feature, you may integrate our IP Phone VIP-153T/VIP-153PT, and the ATA (analog telephone adapter) series - VIP-156/VIP-157 to build up the VoIP network deployment in minutes.

Allowing distributed IP technology to meet traditional voice services, with proactive management interface, the IPX-1803 in the daily business processes, enterprises can make people more productive, more intelligent tasks, and more customer satisfaction.

• SIP features
　- Static/Dynamic registration
　- Call-based MD5 authentication
　- NAT traversal for clients
　- Outbound proxy with or without WAN
　- Up to 5 SIP trunks for inter-PBX SIP trunking
　- Inter-proxy call hand-off
　- Auto-config for SIP IP Phones/ ATA
　- 30 registration / 30 voicemail / 10 concurrent calls
　- Auto NAT Discovery and Traversal
　- Built-in STUN Client
　- RTP Proxy
　- RTP Port Range Designation

• Relational Provision
　- Logical Partition/Relation on Users and Trunks
　- Logical Provision on Outgoing and Incoming Calling Search Scopes
　- Rich Dial-Plan Expressiveness thru Route Patterns
　- User Privilege Propagation Over Intra-Trunks
　- Object-Oriented Provisioning Paradigm

• PBX Features
　- Supports call hold, call waiting, 3-way call conference with feature phones
　- Built-in in-line call transfer
　- Unconditional, unavailable, busy call forward
　- Per-calling-number forward and rejection
　- Call Privilege Grouping
　- Group-based call pick-up
　- Call-parking
　- Up to 8 parties multi-room meet-me conference
　- FXO disconnection tone detection
　- In-band/RFC2833/SIP-INFO DTMF Translation
　- QoS support
　- Lifeline / Emergency call support (IPX-1803)
　- Music on hold
　- Outbound 900/0204 blocking
　- Blacklist of Number Patterns

• Auto Attendant
　- Configurable IVR prompts
　- Key to reach operator
　- Timeout interval and timeout action
　- Music on ringing extensions
　- Forwards to voice mail on no-answer

• Voice Mail
　- User PIN
　- 450 minutes recording time
　- MWI notification
　- E-mail notification
　- Personal reception on unavailability and busy
　- Voicemail forwarding
　- Voicemail to email