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U100 Network IP Telephone Switch

bekiot-U100

U100 VoIP Voice Exchange is a pro-grade converged comm system for mid-large enterprises, supporting 500 SIP extensions, 100 concurrent calls/meetings, 7,500+ hrs recording. Ideal for high-traffic users, it excels in capacity, performance, and UX with rich features to enhance corporate image.......

Product Introduction

U100 VoIP Voice Exchange is a professional-grade converged communication system customized for medium and large enterprises. SIP extension registration capacity is up to 500 channels, concurrent calls can be up to 100 channels, concurrent meetings can be up to 100 parties, and the ultra-long call recording time can be up to more than 7,500 hours. This makes U100 more suitable for enterprise users with huge traffic and high requirements for the performance and stability of IP voice switches. KenePBX-U100 not only performs well in system capacity and performance, but also performs well in user experience. Its rich and diverse enterprise telephone communication functions and extension terminal choices meet the needs of modern office of enterprises and help enhance corporate image and value.
 

Product parameters

500分机
100通话并发
约7500小时录音(本地存储)
管理员用户:具有最高有管理权限
接线员用户:管理分机、高级电话功能,查看传真、录音、通话日志等。
分机用户:网页分机(WebRTC),查看本机录音、留言、通话记录等。
基于 iptables 的防火墙
Geo-IP(基于 IP 地址地理位置的安全策略)
入侵检测和防御
网络模式(WAN):静态IP,DHCP,PPPOE
虚拟专网(VPN):PPTP, OpenVPN, IPSec,L2TP(服务器或客户端)
IVR层级无限制
队列数量无限制
10000 通讯录联系人
50寻呼组成员
10个会议室
500000通话记录(CDR)
1TB USB扩展存储
呼入路由数量无限制
呼出路由数量无限制
开放API接口,可进行二次开发对接第三方系统
IP 白名单/黑名单
分机授信注册 IPSIP(RFC3261), IAX2
DTMF(RFC4733, SIPINFO, In-Band)
传输协议:UDP, TCP, TLS 以及 SRTP
其他网络协议:IPv4, IPv6, VLAN, DHCP, PPPoE, DDNS, NTP, SNTP, TFTP, SSH, HTTPS, LDAP
视频编码(透传):VP8,H.264,H.263+,H.263,H.261
音频编码:Opus, G.722, G.711ulaw, G.711alaw, G.726, G.729, GSM, Speex, AMR, AMR-WB

Featured Features

Quick Setup Wizard
The Quick Installation Wizard is simple, easy to use and powerful, and can quickly guide users to complete the necessary basic settings for new system installation and deployment, without the need for users to have professional knowledge.

Dual machine hot standby
Real-time status monitoring and real-time data synchronization of primary and standby servers. When the primary server fails, the telephone service instantaneously switches to the standby server, which protects the security and stability of enterprise communication.

SIP Proxy Service
No fixed public IP, no third-party dynamic domain name resolution, no VPN router support, remote extension registration, and remote IP telephone system networking problems can be easily solved.

Billing system
No need for the support of third-party billing software, no need for complicated docking settings. Supports prepaid, postpaid, credit limit and other billing methods, and supports flexible rate billing rule definition. Supports billing details and statistical reports.

Web Teleconference Management System
Through the graphical Web interface to realize the conference management function, users can easily initiate 15-party conference calls, and can ban, invite and kick people on the conference.