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Planet Technology IPX-1800N |
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Hardware > Comunicaciones >
Centralitas > Sistema-IP > Planet-Technology >
IPX-1800N
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Internet Telephony PBX System with ISDN Support (30 user registrations, 4 x ISDN) |
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Garantía del fabricante: 2 años SAT (Servicio Técnico) |
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Hoja de Datos |
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Disponibilidad: |
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Características |
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PLANET IPX-1800N ISDN IP PBX system is designed and optimized for the SMBand SOHO daily communications. The IPX-1800N is the next generation voice communication platform for small to medium enterprises. Designed as an open, scalable, and highly reliable telephony solution, the IPX-1800N is able to accept 30 extension registrations, and effectively scales from under 30 users to as many as 50-user enterprises. Designed to run on a variety of VoIP applications, the IPX-1800N provides centralized call control, auto-attendant, voice conferencing, ISDN access, digital and IP-based communications. The IPX-1800N integrates 4 ISDN telephony interfaces to become a feature-rich PBX system that supports seamless communications between existing local calls, IP phones and SIP-based endpoints.
The IPX-1800N 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-1800N, standard SIP phones can be easily integrated in your office; plus the auto-config feature, you may integrate PLANET’s IP Phone VIP-154T series, VIP-155PT, and the ATA (analog telephone adapter) series - VIP-156 / VIP-157 to build up the VoIP network deployment in minutes.
The IPX-1800N in the daily business processes is allowed to distribute IP technology to meet traditional voice services with proactive management interface so that enterprises can make people more productive, make more intelligent tasks, and gain more satisfaction of customers.
• 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 between 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
Support 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
Music on hold
Outbound 900/0204 blocking
Blacklist of Number Patterns
• Auto Attendant
Configurable IVR prompts
Key to reach operator
Work time IVR
3-layer Auto Attendant
Timeout interval and timeout action
Music on ringing extensions
Forward 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 |
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Especificaciones |
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Hardware
LAN 4 RJ-45 (10/100Base-TX, Auto-Negotiation)
WAN 1 RJ-45 (10/100Base-TX, Auto-Negotiation)
Telephony Port 4 RJ-45 (Euro-ISDN ST-interface)
Standards and protocol
Call Control SIP 2.0 (RFC3261)
Registration Max. 30 nodes / SIP IP phones
Calls Max. 10 concurrent calls
Voice CODEC Support G.711, G.726, GSM, G.723.1 (5.3, 6.3kbps), G.729A (8kbps)
Voice Processing
Voice Active Detection,
DTMF detection/generation,
G.165/G.168 echo cancellation (ECN) (25 ms.),
Comfort noise generation (CNG),
Gain Control
PBX Features
Support call hold, call waiting, 3-way call conference with feature phones (VIP-154T series, VIP-155PT and ATA series)
Built-in in-line call transfer
Unconditional, unavailable, busy call forward
Per-calling-number forward and rejection
Group-based call pick-up
Call-parking
Inter-PBX SIP trunking
Multi-room meet-me conference
Auto-attendant
Voice mail system
Call privilege grouping
ISDN interface for PSTN Inbound/outbound
ISDN disconnection tone detection
ISDN hunt group
Caller ID detection
Echo cancellation
In-band / RFC2833 / SIP-INFO DTMF translation
Music on hold
Direct line Outbound
900 / 0204 blocking
Auto Attendant Features
Configurable IVR prompts
Work time IVR
3-layer Auto Attendant
Key to reach operator
Timeout interval and timeout action
Music on ringing extensions
Forward to voice mail on no-answer
Voicemail Features
User PIN
Multilingual
Multi-folder archive
Fast-forward, Rewind, Undelete
MWI notification
E-mail notification and attachment (Unified messaging)
Personal reception on unavailability and busy
Voicemail forwarding
Voicemail to email
Reply calls or new calls in voicemail menu
Internet Sharing
Protocol TCP/IP, NAT, DHCP, HTTPs, DNS
Advanced Function QoS
Management HTTP
Environment
Operating Temperature 0~50 Degree C, 10~90% humidity
Power Requirement 12V DC
EMC/EMI CE, FCC
Network and Configuration
Connection type Static IP, PPPoE, DHCP
Management HTTPs (LAN / WAN access), HTTP (LAN access) |
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