Intuitive, Ease-of-Use IP PBX Management 

PLANET IPX-1100 advanced IP PBX telephony system is effortlessly set up and managed, thanks to its intuitive web-based user interface and quick setup wizard. As an Asterisk-based solution, it offers the complete benefits of pre-loaded SIP, akin to features found in other high-end enterprise-level appliances. With a capacity to accommodate 100 user registrations, the IPX-1100 simplifies the administration of a comprehensive voice-over-IP system, providing both convenience and cost advantages. The equipment is best paired with PLANET color IP phones to get the function going smoothly and to have a seamless connection between the software and hardware.

Model	IPX-1100	IPX-1102
Extension User	100	100
Concurrent Call	50	50
Room Concurrent Call	30	30
Recording/Voicemail	400 hrs	400 hrs
Module	-	2 FXO (Built-in)

Off-net Calling Capability, Call Restriction, Call Access Control

The IPX-1100 is instrumental in establishing a robust VoIP system for small- and medium-sized businesses (SMBs). When seamlessly integrated with PLANET VoIP gateways (VGW-series), the IPX-1100 extends support for analog connections. This integration guarantees seamless communication encompassing the existing PSTN calls, analog phones, IP phones, and SIP-based endpoints.

Distributed VoIP Network Infrastructure

In the new-generation communication age, the IPX-1100 supports IPv6 and VPN (client/server) connection to provide users with more flexible and advantageous communications products. With PLANET DDNS function, the IPX-1100 also helps users to apply and remember the login information easier. Its multiple-language feature helps user to quickly and friendly manage the system. Moreover, the IPX-1100 supports Lync server to which smart phone (using third-party app) and analog phone are connected via its communication with other devices of Lync server.

 

Standard Compliance

Compliant with the Session Initiation Protocol 2.0 (RFC 3261), the IPX-1100 is able to broadly interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide their customers with better multi-media exchange services.

System Highlights

• 50 concurrent calls and up to 100 registers
• 30 conference attendees 
• 400-hour recording (internal storage) 
• Unlimited SIP/IMS trunks
• HD voice codec G.722 for perfect voice quality
• Voicemail to Email for not missing any important message
• Paging and intercom function strengthens the work efficiency
• Built-in SIP proxy server following RFC 3261
• Multiple language of GUI for international business
• Web-based control panel for easy configuration and management of the system
• Hardware echo cancellation module for great and smooth communication
• Strong security features protect your system from hacking
• Records voice and voicemail to external USB disk 
• Quick setup wizard 

Codec and Protocol

• SIP 2.0 (RFC3261), IAX2 and Lync server compliant
• Audio Codec: G.711-Ulaw, G.711-Alaw, G.722, G.726, G.729, GSM, SPEEX, Opus, AMR, AMR-WB 
• Video Codec: H.261, H.263, H.263+, H.264 and VP8
• DTMF: RFC 4733, SIP info, in-band and auto

Network and Security Features

• DHCP server, DDNS client (PLANET DDNS & Easy DDNS)
• SNMP v1/v2, IEEE802.1Q VLAN
• IPv4/IPv6, TR069
• Manual configuration of static route table
• Troubleshooting (Ping and Traceroute)
• VPN server and VPN client
• Mitigates SIP Register DoS attacks 
• Prevents Abort Invite DoS attacks
• Prevents SSH Login DoS attacks
• Firewall and enhances HTTPS connection
• Geo-IP (Security policy based on IP address geographical locations)
• Data backup and recovery

PBX Features

• SIP Register with UDP/TCP/TLS/SRTP
• One Touch Recording
• Mobility Extension
• Black List
• BLF (Busy Lamp Field)
• CDR (Call Detailed Record)
• Conference Room 
• DID (Direct Inward Dialing Number)
• SRTP (Secure Realtime Transport Protocol)
• DND (Do Not Disturb)
• IVR (Interactive Voice Responses)
• Follow Me, Call Spy and PIN Set
• Distinctive Ringtone
• Multi-language System Prompt
• Phone Book, Speed Dial
• Ring Group, SIP Trunk
• Skype for SIP, Smart DID, System Log
• T.38 fax (pass-through), voicemail and voicemail to e-mail

Call Features

• Call Back, Call Forward, Call Group
• Call Hold, Call Paging and Intercom
• Call Park, Call Pickup, Call Queue
• Call Record, Call Route, Blind Transfer 
• Attend Transfer, Call Waiting
• Caller ID, Dial by Name 
• Customized IVR, On-hold Music, Transfer 
• 30 Conference attendees 
• One-to-One Video Call 

 

Hardware
WAN	1 x 10/100BASE-T RJ45 for WAN, connecting to broadband modem or a WAN router
LAN	1 x 10/100BASE-T RJ45 for LAN, connecting to a LAN switch
USB	1 Port for external storage device (up to 2T Bytes) File system format：FAT16, FAT32, EXTFAT, NTFS, EXT3, EXT4
Console Port	1 Port (Baud Rate: 115200)
Reset Key	Press and hold for more than 5 secs to initiate a system reset to default settings
Protocols and Standard
Standard	SIP 2.0 (RFC 3261), IAX2
Protocols	RFC 768 UDP RFC 793 TCP RFC 5246 TLS RFC 3711 SRTP RFC 826 ARP RFC 1034, 1035 DNS RFC 1631 NAT RFC 2068 HTTP RFC 2131 DHCP RFC 2516 PPPoE 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 IPv4, IPv6, VLAN, DHCP, PPPoE, DDNS, NTP, SNTP, TFTP, SSH, HTTPS, LDAP
Audio Codecs	G.711-Ulaw, G.711-Alaw, G.722, G.726, G.729, GSM, SPEEX, Opus, AMR and AMR-WB
Video Codec	H.261, H.263, H.263+, H.264 and VP8
Fax over IP	T.38 Fax (pass-through) Note: T.38 support is dependent on fax machine, SIP provider and network, transport resilience
Voice Processing	DTMF detection and generation RFC 4733, SIP info, in-band and Auto
Internet Sharing
Network Features	DDNS client (Planet DDNS and easy DDNS) DHCP server/SNMP v1/v2 IEEE 802.1Q VLAN IP assignment (DHCP/Static) IPv4/IPv6 Manual configuration of static route table Troubleshooting (ping and traceroute) VPN server and VPN client
Security Features	Mitigates SIP Register DoS Attacks Prevents Abort Invite DoS Attacks Prevents SSH Login DoS Attacks Firewall and enhances HTTPS connection
Features
PBX Features	SIP Register with UDP/TCP/TLS/SRTP Phone Auto-Provision One Touch Recording Mobility Extension Black List BLF (Busy Lamp Field) CDR (Call Detailed Record) Conference Room DID (Direct Inward Dialing Number) DISA (Direct Inward System Access) DNIS (Dialed Number Identification Service) SRTP (Secure Realtime Transport Protocol) DND (Do Not Disturb) FOP (Flash Operation Panel) Status Monitoring IVR (Interactive Voice Responses) Follow Me, Call Spy and PIN Set Distinctive Ringtone Multi-language System Prompt Multiple Language of GUI Phone Book, Speed Dial LDAP Server for phonebook Record Files Download Ring Group, SIP Trunk Skype for SIP, Smart DID, System Log T.38 fax (pass-through), voicemail and voicemail to e-mail Time-based Rule PBX log, web access log and PBX debug log
Call Features	Call Back, Call Forward, Call Group Call Hold, Call Paging and Intercom Call Park, Call Pickup, Call Queue Call Record, Call Route, Blind Transfer Attend Transfer, Call Waiting Caller ID, Dial by Name Customized IVR, On-hold Music, Transfer 30 Conference attendees One-on-One Video Call
System Capacity
System Capacity	50 simultaneous calls Up to 100 IP phone registers/extensions Recording and Voicemail: 400 hours
Network and Configuration
Access Mode	Static IP, DHCP
LED Indications	PWR: 1, LNK (Green) SYS: 1, LNK/ACT (Green) WAN: 1, LNK/ACT (Green) LAN: 1, LNK/ACT (Green)
Dimensions (W x D x H)	167.6 x 115 x 28.8 mm
Operating Environment	0~40 degrees C, 5~95% humidity
Power Requirements	DC 12V, 1A
EMC/EMI	CE, FCC Class B, RoHS