IPX-1800N

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更新時(shí)間：2022-03-14 21:13:29瀏覽次數(shù)：348次

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模擬電話適配器

IP PBX

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深圳市錦辰科技有限公司

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產(chǎn)品簡介

PLANETIPX-1800NISDNIPPBXsystemisdesignedandoptimizedfortheSMBandSOHOdailycommunications

詳細(xì)介紹

PLANET IPX-1800N ISDN IP PBX system is designed and optimized for the SMB and 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, voic, and a widely 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-1800N, standard SIP phones can be easily integrated in your office, plus the auto-config feature, which may be integrated with PLANET?s IP Phone VIP-154T series, VIP-155PT, and the ATA (analog telephone adapter) series - VIP-156 / VIP-157 to build 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 customer satisfaction.

IPX-1800N Intra Office Voice Communication

SIP features

Static/Dynamic registrationCall-based MD5 authenticationNAT traversal for clientsOutbound proxy with or without WANUp to 5 SIP trunks for inter-PBX SIP trunkingInter-proxy call hand-offAuto-config for SIP IP Phones/ ATA30 registration / 30 voic / 10 concurrent callsAuto NAT Discovery and TraversalBuilt-in STUN ClientRTP ProxyRTP Port Range Designation
Relational Provision

Logical Partition/Relation between Users and TrunksLogical Provision on Outgoing and Incoming Calling Search ScopesRich Dial-Plan Expressiveness thru Route PatternsUser Privilege Propagation Over Intra-TrunksObject-Oriented Provisioning Paradigm
PBX Features

Support call hold, call waiting, 3-way call conference with feature phonesBuilt-in in-line call transferUnconditional, unavailable, busy call forwardPer-calling-number forward and rejectionCall Privilege GroupingGroup-based call pick-upCall-parkingUp to 8 parties multi-room meet-me conferenceFXO disconnection tone detectionIn-band/RFC2833/SIP-INFO DTMF TranslationQoS supportLifeline / Emergency call supportMusic on holdOutbound 900/0204 900/0204 FREE blockingBlacklist of Number Patterns
Auto Attendant

Configurable IVR promptsKey to reach operatorWork time IVR3-layer Auto AttendantTimeout interval and timeout actionMusic on ringing extensionsForward to voic on no-answer
Voic

User PIN450 minutes recording timeMWI notification notificationPersonal reception on unavailability and busyVoic forwardingVoic to

硬件特性
LAN	4 RJ-45 (10/100Base-TX, Auto-Negotiation)
WAN	1 RJ-45 (10/100Base-TX, Auto-Negotiation)
電話端口	4 RJ-45 (Euro-ISDN ST-interface)
標(biāo)準(zhǔn)和協(xié)議
呼叫控制	SIP 2.0 (RFC3261)
注冊	Max. 30 nodes / SIP IP phones
來電	Max. 10 concurrent calls
支持語音 CODEC	G.711, G.726, GSM, G.723.1 (5.3, 6.3kbps), G.729A (8kbps)
語音處理	Voice Active Detection, DTMF detection/generation, G.165/G.168 echo cancellation (ECN) (25 ms.), Comfort noise generation (CNG), Gain Control
PBX功能	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 Voic 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
自動(dòng)話務(wù)員功能	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 voic on no-answer
語音信箱功能	User PIN Multilingual Multi-folder archive Fast-forward, Rewind, Undelete MWI notification notification and attachment (Unified messaging) Personal reception on unavailability and busy Voic forwarding Voic to Reply calls or new calls in voic menu
互聯(lián)網(wǎng)共享
協(xié)議	TCP/IP, NAT, DHCP, HTTPs, DNS
高級功能	QoS
管理	HTTP
環(huán)境
操作	Temperature 0~50 degrees C, 10~90% humidity
電源需求	12V DC
EMC/EMI	CE, FCC
網(wǎng)絡(luò)配置
連接類型	Static IP, PPPoE, DHCP
管理	HTTPs (LAN / WAN access), HTTP (LAN access)

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