MSME-Verified Enterprise Open Source PBX Engine

Next-Generation Open Source
VoIP & PBX Telecom Infrastructure.
Carrier-Grade Asterisk Telemetry!

Carrier-Grade Asterisk SIP Server, Real-Time CDR Analytics, High-Availability Extension Management, And Encrypted Voicemail Routing Infrastructure Developed By Durgaai Solutions.

cell_tower Dual PJSIP Transports (UDP & TLS)
history_toggle_off Real-Time Forensic CDR Logging
qr_code_scanner Instant Line Allocation Via KYC QR
verified_user 100% Encrypted TLS 1.3 & SRTP
explore Explore Key Capabilities terminal Try Interactive VoIP Sandbox
PBX Server
ONLINE
AMI Socket 5038 Active
System Uptime
00h 00m 00s
Continuous Operational
Active Channels
0
Live Bridge Calls
PJSIP Signaling
UDP & TLS
SIP Port 5060 Enabled
Audit Logger
Master.csv
Forensic Record Parser
Voicemail Spool
ACTIVE
Routing (*ext)
// KEY CAPABILITIES

Enterprise Solutions & Key Capabilities

Advanced Telecommunications Engineering Powered By Open Source Asterisk Technology

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phone_in_talk

PJSIP Endpoint Provisioning

Automated PJSIP Configuration Generator Supporting Symmetric RTP, NAT Traversal, Multi-Contact Binding, And Custom Expiration Timers For Enterprise Extensions.

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query_stats

Real-Time CDR Forensic Analytics

Comprehensive Call Detail Record Logging With Instant Parsing, Timestamp Tracking, Source And Destination Auditing, And Call Duration Analytics.

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voicemail

Encrypted Voicemail Spooling

Direct Mailbox Shortcuts (*ext), Custom PIN Access, MP3 Greeting Audio Support, And Automated Email Notification Routing For All Extensions.

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dns

Dockerized High Availability Engine

Isolated Docker Container Architecture With Automated Startup Delay Routines, Health Check Execution, And Asterisk Manager Interface Core Reloading.

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verified_user

Granular Access Control & KYC Verification

Instant User Suspension, Extension Revocation, Mailbox Toggling, Document ID Verification, And Administrative Audit Logging.

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devices

Cross-Platform Softphone Client Compatibility

Out-Of-The-Box Support For Zoiper, MicroSIP, Linphone, MizuDroid, GS Wave, And Hardware IP Phone Provisioning.

// ARCHITECTURE BLUEPRINT

Telecom Systems Architecture & Engineering Blueprint

Comprehensive Technical Breakdown Of The Durgonix VoIP Telecommunication Infrastructure Built By Durgaai Solutions

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hub

Asterisk PJSIP Master Core & Channel Engine

Native C-Based PJSIP Driver Orchestrating Real-Time SIP Registrations, In-Dialog Re-Invites, And Multi-Contact AOR Binding With Zero Packet Loss Under Heavy Concurrent Loads.

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description

Forensic Master CDR Processing Pipeline

Automated Parsing Engine Ingestion Stream Real-Time Forensic Logs From /var/log/asterisk/cdr-csv/Master.csv With Nanosecond Precision, Source-Destination Mapping, And Disposition Auditing.

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lock

Dual Transport Security & TLS Encryption

Simultaneous Dual-Stack Transport Supporting Standard UDP Port 5060 Alongside Encrypted TLS 1.3 Port 50826 With SRTP Media Protection To Guarantee Confidential Enterprise Voice Communications.

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badge

Automated Subscriber KYC & Line Allocation Certification

Instant Automated Generation Of Cryptographic Line Certificates Bearing Verifiable SHA-256 Hashes, Official Digital Signatures, And QR Verification Codes For Legal Telecom Compliance.

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settings_voice

Direct Voicemail Spooling & Audio Transcoding

Dedicated Mailbox Routing Engine (*ext) With PIN Security, Dynamic Audio Transcoding Across ULAW, ALAW, And GSM Codecs, And Automated Real-Time Voicemail Indicator Synchronization.

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sync_alt

Ecosystem Interoperability (SSTA Walkie Talkie & Synrix Forge)

Native Integration With Durgaai Solutions SSTA Walkie Talkie For Seamless Voice Broadcast Relays, And Synrix Forge For Immutable Version Control Of Telecom Infrastructure Configurations.

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// TECHNICAL TELEMETRY MATRIX

Infrastructure Protocol & Deployment Standards

Status: Verified Operational
Signaling Standards
SIP 2.0 (RFC 3261) • PJSIP

Compliant With RFC 3261 Signaling, Symmetric RTP Handshakes, And Dynamic SDP Negotiation.

Transport Gateways
UDP 5060 / 50828 • TLS 50826

Dual-Channel Listener Supporting Low-Overhead UDP And High-Security Mutual TLS 1.3.

Carrier Audio Codecs
G.711u • G.711a • G.722 • Opus

Ultra-Low Bitrate Compression Delivering Studio-Grade HD Voice Clarity Over Low Bandwidth.

Management & Forensics
AMI Socket 5038 • Master.csv

Asterisk Manager Interface TCP Socket Paired With Comprehensive CDR Forensic Data Logging.

// INTERACTIVE SIMULATOR

Interactive Extension Routing Simulator

Test Extension Dialing Logic And Direct Voicemail Routing Rules Live In Your Browser

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Preset Extensions:
[System Ready] Interactive Extension Routing Simulator Initialized.
[Instruction] Select A Preset Extension Or Enter An Extension ID Above And Click A Simulation Mode.
// CLIENT PROVISIONING

Softphone Client Setup & Configuration Guides

Step-By-Step Instructions For Connecting Mobile And Desktop Softphones To The PBX Server

1

Download And Install Zoiper Softphone

Download The Official Zoiper Application For Windows, macOS, Android, Or iOS From Your Preferred Application Store.

2

Add A New SIP Account

Open Zoiper Settings, Navigate To Accounts, Tap Add Account, And Select Manual SIP Account Configuration.

3

Configure SIP Credentials And Registrar Host

Set Account Name To Your Extension ID (e.g. 1608), Password To Your Secret SIP Password, And Domain / Host To 176.31.163.114:50828.

// ARCHITECTURE SPECS

System Documentation & Architecture Specification

Inspect Open Source Asterisk Dialplan Configurations, AMI Command Flows, And KYC Certificate Generation Rules

1. PJSIP Custom Dialplan Template

Every Provisioned Subscriber Line Generates Dynamic PJSIP Endpoints With Context Internal, Symmetric RTP, Direct Media Bypassing, And Custom Mailboxes.

2. Master CDR CSV Audit Logging

Calls Are Logged To /var/log/asterisk/cdr-csv/Master.csv For Real-Time Forensic Analysis, Source/Destination Tracking, And Duration Auditing.

3. Line Allocation KYC Certificate

Generate And Print Official KYC Verification Certificates Containing Subscriber Credentials, KYC Serial Numbers, And SIP Softphone Instructions.

// FAQS

Frequently Asked Questions

Common Technical Questions Regarding The Durgonix PBX Telecom Platform

Durgonix Telecom Is An Enterprise-Grade Open Source PBX Telecom Platform Developed By Durgaai Solutions. It Provides Automated PJSIP Extension Provisioning, Real-Time Call Detail Record (CDR) Audit Logging, Direct Voicemail Routing, And Full Asterisk Container Management.
After An Administrator Provisions A SIP Extension In The Admin Operator Panel, You Can Connect Softphone Apps Like Zoiper, MicroSIP, Or Linphone Using Your Extension ID, Secret SIP Password, And Server Domain Port 5060 / 50828.
Each Extension Has A Direct Voicemail Shortcut. For Instance, Dialing *1608 Routes Directly To The Mailbox PIN Prompt For Extension 1608, Allowing Users To Access Mailbox Audio Or Leave Messages Instantly.
Call Logs Are Written Automatically To Asterisk's Master.csv File. The Admin Operator Panel Parses This File In Real Time, Enabling Forensic Search, Extension Filtering, Duration Breakdown, And Record Clearing.
Yes! The VoIP Server Is Containerized Using Docker And Docker Compose, Featuring Automated Container Health Checks, Host Bridge Networking, Certificate Generation Scripts, And AMI Core Hot-Reloading.
Click The Prominent Login To Admin Portal Button Located In The Top Navigation Bar Or CTA Banner, Enter Your Registrar ID And Admin Access Code, And Click Initialize Session.
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Ready To Provision Your Enterprise PBX System?

Access The Durgonix Telecom Operator Panel To Provision Extensions, Audit Call Logs, And Manage Voicemail Accounts.