Chapter 1 vip-2100 introduction, System description – Planet Technology VIP-2100 User Manual
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VIP-2100 User
’s manual
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Chapter 1 VIP-2100 Introduction
System Description
VIP-2100 is a cost effective solution for VoIP trunk gateway supporting one-,
port T1/E1 VoIP trunks that provides voice and fax over IP network. It
supports ITU-T H.323 V3, SIP RFC 2543/3261, SNMP V2, Call Detail Record,
WEB management and other useful functions to meet customer requirements.
The built-in enhanced IVR (Interactive Voice Response) and Billing Service
of VIP-2100 is suitable for prepaid and postpaid service. It can rapidly provide
value added service for customers.
VIP-2100 Features:
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Dual SIP/H.323 co-existing
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ITU-T H.323 v3 and H.450 compliance
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SIP RFC 2543/3261 standard compliance
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PSTN signaling: ISDN/PRI, CAS (MFC R2, MFC R1, E&M), QSIQ
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Mixed SIP, Gatekeeper and P2P calls
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Support H.323 Gatekeeper register, direct and route calls
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Support SIP outbound proxy, redirect and register server
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Redundant SIP Proxy/Outbounbd Proxy Server Support (Outbond
Active/Active fail over, Register A/A no fail over)
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Support SIP Overload Redirect
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SIP supplemental service - on Hold, Call Transfer (Transferred)
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Built-in phone book and prefix routing for SIP and H.323 P2P calls
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Support H.323 fast connect, early H.245 and H.245 tunneling
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Support H.323 and SIP early media
- VoIP to VoIP calls support
– SIP to H.323, SIP to SIP, H.323 to H.323
- Global Trunk-Channel Block out: 0xffffffff (busy block out)
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Intelligent PSTN call routing and in-trunk hunting: reverse rotary, channel
mask (default:0xffffffff), ANI prefix match
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Reset a channel/trunk on the fly
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Flexible digit manipulation plan
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Support RADIUS Authentication, Authorization and Accounting
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Support access control by ANI, DNIS, IP, Gatekeeper only, proxy only or
RADIUS
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SIP UDP/TCP support
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Behind NAT friendly for SIP calls
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Inbound and out of band DTMF transmission
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SIP/H.323 T.38 fax relay up to 14400 bps
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Dynamic call treatment based on DNIS, ANI or collected DTMF
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Grouping DNIS/ANI Number Replacement
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Built-in IVR & call-flow controller for PSTN / VoIP side
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CISCO compatible
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Web-based graphic announcement edit and management