EZPBX-2000 IP-PBX

EZPBX-2000 IPV4+V6 Dual IP-PBX

A cost effective IPV4+IPV6 dual stacks SIP IP-PBX software running under 64 bits Linux.

EZPBX-2000 IP-PBX is an ideal IP-PBX for small to large enterprise which provides very powerful built-in rich telephony service at affordable price. It supports multi-office, multi-languages auto attendant, voice mail and up-to 32 parties conference. It support IPV4 and IPV6 dual stacks simultaneously. Also the hitless redundant feature increases reliability and keep the business running.

With smart calling features, it enables your smart phone to become a part of the office extension. You can use it to create conference, click to call, forward to your IP-PBX calls to your smart phone. You can communicate to your customer no matter where you are. And the voice logging feature enabled you able to extend your PBX into a voice recording system.

Highlights and Benefit

• Support both IPv4 and IPv6 SIP Calls Simultaneously

• Support Hitless Redundancy

• Smart Calling Features for Android and iPhone

• SIP UDP, TCP, TLS Seamless Access

• Automatic Audio/Video NAT Detection and Traversal

• Auto Attendant Service/AA Call Flow Editor

• Voice Mail Service/MWI/Email Notice

• 32 Parties Conference Room/Meeting Me Conference/Dialing Out Conference

• 64 Parties Broadcasting Service

• Support G.711, G.722, G.729, GSM

• Divisional Management & Billing

• Support SIP Trunk and SIP Router

• Flexible yet Powerful Digit Processing and Call Routing Plan

• Easy Web Management and System Morning

• Prosperous Telephony Features for Time to Market

• Detect Potential SIP Attacks and Prevention

• Country IP/Network Lock

• CPE Auto Provisioning

• Support RFC-8599 Push Softphone

• Support SRTP Transcode

• In Call Service

• Optional Smart Calling and Voice Logging Module

• Running under Off-the-Shelf Server and 64 bits Linux

Product Specifications

System Requirements

• INTEL/AMD CPU (Intel® 64)

• RHEL 8/Rocky Linux 8

• MYSQL Database

Protocols

• SIP RFC 3261

• SIP UDP, TCP, TLS

• RTP, SRTP, RTCP

• IPV4 / IPV6 Dual Stack

System Service

• Support Hitless HA Redundant

• Support Automatic TCP and UDP Traversal

• Support Automatic IPV4 and IPV6 Traversal

• Support Multiple SIP Domains

• Automatic Audio/Video NAT Traversal

• SIP Proxy/Registrar

• Support Permanent & Dynamic Contact

• Support SIP Trunking

• Support LAN/WAN Access Simultaneously

• RADIUS Billing Support

• Extension/Device Monitoring

• Device Allowance Control

• Session Timer Call Validation

• INVITE-Initiated Dialog Event Package (RFC 4235)

• External Voice Mail Server Support

• Missed Call Email Notice

• CPE Auto Provisioning

• Support Multi-Office/Branch

• Support RFC-8599 Push Softphone

• Support Flat File/Syslog CDR

• Support SRTP Transcode

Audio Codec

• G.711

• GSM (Full Rate)

• G.722

• G.729

Routing Plan

• Group Based Routing

• Time of Day Routing

• Preference Routing

• Round Robin Routing

• Load Balancing Routing

• Broadcast Routing

• Unavailable Redirect

• ENUM Routing

• ANI Based Routing

• Black List Routing

Telephony Features

• DID/DOD

• Call Transfer

• Call Hold

• Call Waiting

• Call Forward

• Call Display Name

• Call Pickup (Group, Specified, Global)

• Calling Line Identification Presentation (CLIP)

• Calling Line Identification Restriction (CLIR)

• Digit Manipulation

• Local Emergency Call Group

• Secondary PSTN Number

• Parallel Hunting for Multiple Contacts

• Follow Me Always

• Time of Day Follow Me

• Incoming Call Blocking

• Outgoing Call Blocking

• Outgoing Privilege Calling

• Do Not Disturb

• Anonymous Call Blocking

• Camp-On Call

• Call Park/Retrieve

• Music on Hold

• Calling Password Protect

• Distinct Ringing

Broadcasting Service

• Up-to 64 Parties Broadcasting Target

• CPE Auto Answer to Speaker by SIP

• Start/Stop Tone Notice

In Call Service

• During the talk, press ## to start

• Call Flit/Transfer/Hold/Un-Hold

• Support PSTN/Mobile User

Management

• Multi-Language Support

• Web Provisioning Access Log

• Easy Web Management (HTTP/HTTPS)

• 3 Levels of Access Controlling Rights

• Customizable Web Access Right

• SIP Attack Detection and Prevention

• System Alert through SYSLOG and Email

• SOAP Provisioning Interface

• Real-time Status & Tracing

• Scheduled Update Task

• Support Google Authenticator 2FA

System Capabilities

• Max Extensions: 2,000

• Max Concurrent Calls: 1,000

• Max NAT Resource: 1,000

• Max Universal Resource: 256

SIP Attack Detection and Protect

• SIP Attack Detection/IP Blocking

• SIP User Device Restriction

• Country/IP Network Lock

• Enhanced Password Option

• Black Routing List

• CAPTCHA to Protect Web

• Web Access Log

Auto Attendant

• Support Multi-languages

• Support Multiple Offices

• Graphic Attendant Flow Editor

• Incoming Calls Limitation

• Office/DID Based Call Flow

• Up-to 3 Time Segment

• Working/Off-Time/Holiday Operator

• Working/Off-Hours Flow

• Priority/Holiday Flow

• Black List Filter & Flow

• Access to Voice Mail

• Outgoing Calling (Password Protect)

• Access to Meeting Me Conference

Voice Mail

• Support Multi-languages

• Incoming Calls Limitation

• Message Detail

• Voice Mail to Email (MP3)

• Access Voice Mail via Web

• Access Voice Mail via Phone

• SIP MWI (RFC 3842)

• Personal Greeting

Conference

• Up-to 32 Participants Conference

• Support Multi-languages

• Incoming Calls Limitation

• Support Meeting Me Conference

• Hosting/Participant Password

• Join/Quit Announcement

• Support Dialing Out Conference

• Hosting Password

• Dynamic Participant List Building/Calling

• Predefine Participant List Building/Calling

• Join/Quit Announcement

• Unavailable Announcement

• Add Participant within Conference

Voice Logging (optional)

• Max Logging Channels: 512

• Support Selectable Logging Target or Number

• Support G.711, GSM, G.722, G.729 and iBLC decode

• WAV/MP3 Compress with Optional AES encryption

• Separate Caller/Called Voice Channel

• Provides Voice Logging Detail Record

• Support Recording On Demand

Smart Calling Feature (optional)

• Support Android and iPhone

• Forward to Smart Phone

• Click to Call (Call To)

• Create Outgoing Conference

• Monitor Meeting Me Conference

• Conference Control

• Add Participant (Outgoing Conference)

• Remove Participant

• Speak Request

• Mute/un-Mute

Billing Feature

• Support Charge Division

• Top Usage Users Report

• Top Prefix Usage Report

• Prefix Summaries Report

• Division Billing Report

• Division Wide Tariff Plan

• Charge Unit

• Charge Amount

• Call History Detail Report

• Calling/Called Number

• Call Duration

• Call Type

• Call Connect/Disconnect Time

• SIP Call ID

• SIP URI

• Source/Destination IP Address

• Charge Amount

Screenshots

• Creating an Office

• Call Features

• Call History Detail Report

• Auto Attendant Call Flow

• Web Access to Voice Mail

• Voice Logging Report

• Top Prefix Usage Report

• Smart Calling

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