当前位置：深圳市锦辰科技有限公司>>IP语音>>IP PBX>>IPX-2000

IPX-2000

返回列表页

参考价

面议

具体成交价以合同协议为准

产品型号

品牌

厂商性质其他

所在地

在线询价收藏产品

更新时间：2022-03-14 21:14:56浏览次数：596次

联系我时，请告知来自智慧城市网

产品分类 品牌分类

  IP语音

多点控制单元（MCU）

语音网关

模拟电话适配器

IP PBX

全部产品列表

暂无信息

深圳市锦辰科技有限公司

经营模式：其他

商铺产品：408条

所在地区：

联系人：张薇薇

询价 给他留言

产品简介

PLANETIPX-2000isthenextgenerationvoicecommunicationplatformforthesmalltomediumenterprises

详细介绍

PLANET IPX-2000 is the next generation voice communication platform for the small to medium enterprises. Designed as an open, scalable and highly reliable telephony solution, the IPX-2000 is able to accept 200 extension registrations, and effectively scales from under 100 users to as many as 200 in a standard rack-mountable unit. PLANET IPX-2000 is also designed to operate on a variety of VoIP applications; it provides centralized call control, auto-attendant, voice conferencing, and PSTN access, digital and IP-based communications.

The IPX-2000 integrates up to 8 calls via the IPX-FXO (Foreign eXchange Office, FXO) module to become a feature-rich PBX system that supports seamless communications between the existing PSTN calls, analog, IP phones and SIP-based endpoints.

Moreover, the IPX-2000 also integrates telephony call processing, call control, voic, and a wide 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-2000, standard SIP phones can be easily integrated in your office, plus the auto-config feature, which may be integrated with PLANET IP phone (VIP-254 series), and the analog telephone adapter (VIP-156 / 157 series) to build the VoIP network deployment in minutes.

Allowing distributed IP technology to meet the traditional voice services, with the IPX-2000 proactive management interface in the daily business process, it brings enterprises higher employee productivity and customer satisfaction.

Allowing distributed IP technology to meet traditional voice services, with the IPX-2000 proactive management interface in the daily business process, it brings enterprises higher employee productivity and customer satisfaction.

VIP-156 + VIP-351PT IPX-2000 integration

IPX-2000 intra office voice communication

System Highlight

200 max users / extensions with voic account50 max concurrent sessionsHighly integrated, embedded system for stabilityImmediate VoIP connectionsAnalog interface (FXO / FXS) supportFlexible dialing plansOn-ramp / off-ramp calls for VoIP and PSTNRemote managementWizard for easy configurationCommａnd Line Interface (CLI) for quick-batch configurationSeamless integration with legacy PBXAuto provision and auto firmware upgrade for feature IP phones
Feature Highlight

SIP registrar and SIP proxyMulti-language voice prompts for international businessCustomizable 3-layer IVRACD (Automatic Call Distribution) to form basic Call CentersStackable design for scalability and investment protectionUMS (Unified Messaging System) supportMeet-me conference for virtual meeting roomSystem status and system logCall Detail Record (CDR)Built-in STUN clientFunction-rich voic System CAC (Call Admission Control)Call keep alive

硬件特性
接口	One RJ-45 10/100 base-TX WAN Ethernet port One RJ-45 10/100 base-TX LAN Ethernet port Two USB 2.0 ports One RS-232 serial port Two expandable PCI interface slots
标准和协议
注册	Max. 200 nodes / SIP IP phones
来电	Max. 50 concurrent calls
呼叫控制	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, RFC 3264, RFC 3362, RFC 4612
SIP注册	Static / Dynamic registration, Configurable Expiry Time, MD5 authentication Handle loose RFC-compliant SIP devices, Resilient message retry mechanism Cache client registrations
SIP代理	Stateful proxy server, NAT traversal for clients, Inter-proxy call hand-off Outbound Proxy behind NAT Device
PBX 系统
呼叫功能	Codec G.711 (?/A-law), G.723.1 (6.3k/5.3k bit/s), G.729A, and G.726 (16k/24k/32k/40k bit/s) supported Transcoding channel 0~16, subject to add-on card In-bａnd / RFC2833 / SIP-INFO DTMF translation Two expandable slots for telephony interfaces 50 SIP trunks for ITSP account or private trunking shared by extensions 200 DID SIP trunks to extensions Support gateway trunk mode per SIP trunk Enable / Disable NAT Traversal per SIP trunk Call admission control of call count or bandwidth per SIP trunk Long call audit Support SIP OPTIONS keep alive NAT session keep alive Configurable RFC 2833 payload type per SIP trunk FXS / FXO analog trunking FXO disconnection tone detection FXO disconnection tone parameter setting FXS hot line FXS warm line Caller ID detection Trunk hunting Digits manipulation during hunting Life-line priority call Support SIP Call Hold, Call Waiting Support SIP phone 3-way conference Support Blind / Attended Transfer In-line Call Transfer Unconditional, Unavailable, Busy Call forward Call Back on Busy between extensions Per calling number forward and rejection Blacklist of number patterns 32 call pick-up groups Call Park and Retrieve Recording on demａnd with PLANET IP phones Remote extension registration via Internet Direct line to extension (DID to Extension) Direct line by called number (DID by Number) Direct line by privilege (DID by Privilege) Echo Cancellation (G.168) Flexible numbering plan Call privilege grouping Configurable Music on Hold Memo Call for extension Schedule-based Broadcast Support T.38 over IP Support T.30, T.38 pass through ENUM resolution
NAT	Auto NAT discovery and traversal Built-in STUN client RTP proxy RTP port range designation
IVR	50 configurations of 3-layer IVR Worktime / Holiday setting for different IVR Configurable greeting prompts Music on Ringing extensions Forward to Voic on No-answer Support 3 languages in IVR tree Hot key to operator
语音	User Authentication by PIN Multilingual, 3 languages Multi-folder Archive Fast-forward / Rewind / Undelete MWI notification VMWI notification notification and attachment (Unified messaging) Personal greeting on unavailability and busy Record personal greeting through phone Voic Forwarding Reply call or new call after logged in Voic menu Built-in 40GB hard disk drive for Voic Support USB 2.0 interface for Voic, CDR, and system configuration backup Support NFS remote backup for Voic, CDR, and system configuration
会议	24 conference rooms with configurable number and PIN Up to 24 parties among all conference rooms Lock / Mute / Join / Drop control for administrator Music on First Dial-in Party Hot key to leave the conference Hot key for administrator to manage the conference
自动呼叫分配	32 queues with 32 agents among all queues. 32 inbound call among all queues Configurable waiting length for individual queue Support five distribution policies including round robin, ring all, least recent, fewest call, and random Configurable waiting time for each queue Allow agent remotely log-in Agent can participate multiple queues Agent phones also allow extension calls
可堆叠	Support LAN stacking up to 4 units in the same model Automatic intra-trunking creation among stacking units Automatic configuration publishing from Master to Slaves Automatic load balancing in hosting feature phones3
互联网共享
系统管理	Web-based configuration with session control User and administrator configuration mode Automatic expiring the idle sessions Support firmware upgrade through the Internet Configuration Wizard for mass extensions and users creation Step-by-Step Wizard for adding users, extensions and trunks Built in online help in wizard Commａnd Line Interface (CLI) for configuration System event Syslog
网络管理	DHCP / PPPoE / Static IP on WAN Support MAC Clone on WAN Static LAN routing Firewall on predefined services Virtual Server for client device NAT for outbound traffic from LAN WAN QoS queuing mechanism for VoIP and data traffic Support TOS setting DNS forwarder and dynamic DNS SNMPv2 with standard MIB format Adaptive WAN bandwidth and DSP channel saving
环境
工作温度	Operating temperature 0~50? Storage temperature -10~70? Humidity (RH) 10~80% non-condensing
电源要求	100~240V AC, 50~60 Hz
EMC / EMI	CE, FCC