VoIP Engineer for Asterisk/FreePBX Setup & Guide
Budget: €750 – €1,500 EUR
Project Description:
We are looking for an experienced VoIP engineer to set up Asterisk with FreePBX on our Linux-based server. The setup should be fully configured for inbound and outbound calls, include documentation for number management, and provide detailed instructions on how we can extract call event data (Call Start, Call End, Duration, Caller ID, Destination, etc.) from Asterisk or FreePBX if we decide to integrate it with an external API in the future.
Rather than building the API itself, the freelancer should document the methods available within Asterisk/FreePBX to capture and extract call data for further use.
The engineer will be required to extensively communicate with our developer team to gather the necessary information and resources, and ensure the setup aligns with our requirements and a faster delivery is ensured.
Project Scope & Requirements
1. Asterisk & FreePBX Installation & Configuration
• Install and configure Asterisk and FreePBX on a Linux-based server.
• Configure SIP trunks, extensions, IVR, call routing, voicemail, and call recording.
• Implement outbound dial plans, DID routing, and fallback mechanisms.
2. Number Configuration & Routing
• Provide a detailed guide for managing numbers within FreePBX.
• Configure DID (Direct Inward Dialing) numbers, call forwarding, and caller ID settings.
• Ensure proper failover settings for call routing.
3. Call Data Extraction – Guide on How to Retrieve Call Parameters
• Instead of building an API, the freelancer should provide detailed documentation on:
• How to extract call event data from Asterisk/FreePBX in real-time or after a call.
• Where call logs and CDR (Call Detail Records) are stored and how to access them.
• How to configure Asterisk/FreePBX to log call events with relevant details (Caller ID, Call Start, Call End, Call Duration, Destination, Call Status, etc.).
• Methods to export or process call data, including:
• Using Asterisk Manager Interface (AMI) to retrieve call events.
• Using Asterisk Call Detail Records (CDR) logs for historical data.
• Using Asterisk AGI (Asterisk Gateway Interface) for call event handling.
• Using FreePBX’s built-in CDR module for extracting call logs.
• How we could later send extracted data to an external API, if required.
• Provide examples of extracting call data and modifying the output format (CSV, JSON, database storage, etc.).
• Guide on how to customize event triggers (e.g., capturing specific call types, ignoring internal calls).
4. Security & Performance Optimization
• Implement firewall rules, Fail2Ban, and secure SIP trunk settings.
• Optimize call quality (jitter buffer, codecs, and latency management).
• Prevent unauthorized SIP registrations and ensure system hardening.
5. Comprehensive Documentation (Step-by-Step Guide)
The documentation should be written for non-experts and cover:
• FreePBX GUI instructions for managing numbers, trunks, and extensions.
• Asterisk CLI commands for debugging and administration.
• Call data extraction methods, including:
• How to access CDR logs.
• How to use AMI for real-time call event tracking.
• How to configure Asterisk to log custom parameters.
• Example configurations for exporting data.
• How to adapt the output format for future API integration.
• Security best practices for ongoing maintenance.
6. Testing & Handover
• Perform test calls (inbound & outbound) to ensure full system functionality.
• Demonstrate how call data is logged and accessed.
• Provide basic post-installation support for bug fixes and questions.
Ideal Freelancer Profile
• Extensive experience with Asterisk & FreePBX setup.
• Strong knowledge of VoIP security, SIP, and call logging.
• Ability to write clear, step-by-step documentation.
• Experience with Asterisk AMI, AGI, and CDR logs.
• Fluent in English for effective communication.
• Willing to work closely with our developer team to ensure a smooth implementation.
Deliverables
• Fully installed and configured Asterisk + FreePBX.
• Secure VoIP setup with working SIP trunks and call routing.
• Step-by-step documentation for FreePBX management & call data extraction.
• Guide on how to retrieve call parameters from Asterisk/FreePBX.
• Tested system with call event logging demonstrated.
• Initial troubleshooting support.
We are looking for an experienced VoIP engineer to set up Asterisk with FreePBX on our Linux-based server. The setup should be fully configured for inbound and outbound calls, include documentation for number management, and provide detailed instructions on how we can extract call event data (Call Start, Call End, Duration, Caller ID, Destination, etc.) from Asterisk or FreePBX if we decide to integrate it with an external API in the future.
Rather than building the API itself, the freelancer should document the methods available within Asterisk/FreePBX to capture and extract call data for further use.
The engineer will be required to extensively communicate with our developer team to gather the necessary information and resources, and ensure the setup aligns with our requirements and a faster delivery is ensured.
Project Scope & Requirements
1. Asterisk & FreePBX Installation & Configuration
• Install and configure Asterisk and FreePBX on a Linux-based server.
• Configure SIP trunks, extensions, IVR, call routing, voicemail, and call recording.
• Implement outbound dial plans, DID routing, and fallback mechanisms.
2. Number Configuration & Routing
• Provide a detailed guide for managing numbers within FreePBX.
• Configure DID (Direct Inward Dialing) numbers, call forwarding, and caller ID settings.
• Ensure proper failover settings for call routing.
3. Call Data Extraction – Guide on How to Retrieve Call Parameters
• Instead of building an API, the freelancer should provide detailed documentation on:
• How to extract call event data from Asterisk/FreePBX in real-time or after a call.
• Where call logs and CDR (Call Detail Records) are stored and how to access them.
• How to configure Asterisk/FreePBX to log call events with relevant details (Caller ID, Call Start, Call End, Call Duration, Destination, Call Status, etc.).
• Methods to export or process call data, including:
• Using Asterisk Manager Interface (AMI) to retrieve call events.
• Using Asterisk Call Detail Records (CDR) logs for historical data.
• Using Asterisk AGI (Asterisk Gateway Interface) for call event handling.
• Using FreePBX’s built-in CDR module for extracting call logs.
• How we could later send extracted data to an external API, if required.
• Provide examples of extracting call data and modifying the output format (CSV, JSON, database storage, etc.).
• Guide on how to customize event triggers (e.g., capturing specific call types, ignoring internal calls).
4. Security & Performance Optimization
• Implement firewall rules, Fail2Ban, and secure SIP trunk settings.
• Optimize call quality (jitter buffer, codecs, and latency management).
• Prevent unauthorized SIP registrations and ensure system hardening.
5. Comprehensive Documentation (Step-by-Step Guide)
The documentation should be written for non-experts and cover:
• FreePBX GUI instructions for managing numbers, trunks, and extensions.
• Asterisk CLI commands for debugging and administration.
• Call data extraction methods, including:
• How to access CDR logs.
• How to use AMI for real-time call event tracking.
• How to configure Asterisk to log custom parameters.
• Example configurations for exporting data.
• How to adapt the output format for future API integration.
• Security best practices for ongoing maintenance.
6. Testing & Handover
• Perform test calls (inbound & outbound) to ensure full system functionality.
• Demonstrate how call data is logged and accessed.
• Provide basic post-installation support for bug fixes and questions.
Ideal Freelancer Profile
• Extensive experience with Asterisk & FreePBX setup.
• Strong knowledge of VoIP security, SIP, and call logging.
• Ability to write clear, step-by-step documentation.
• Experience with Asterisk AMI, AGI, and CDR logs.
• Fluent in English for effective communication.
• Willing to work closely with our developer team to ensure a smooth implementation.
Deliverables
• Fully installed and configured Asterisk + FreePBX.
• Secure VoIP setup with working SIP trunks and call routing.
• Step-by-step documentation for FreePBX management & call data extraction.
• Guide on how to retrieve call parameters from Asterisk/FreePBX.
• Tested system with call event logging demonstrated.
• Initial troubleshooting support.