FreePBX Install & Call API

@ahmadayaz

I am an experienced FreePBX and Asterisk developer who understands your need to automate telephone bridge calls via a REST API. I plan to utilize FreePBX for easier management and configuration flexibility, integrating multiple SIP trunks and enabling automatic call recordings for enhanced security, stability, and scalability. I will develop a REST API with call bridging functionality, real-time event notifications via webhooks, and automated call recording uploads. The system will support multiple projects independently, ensuring complete isolation and customized configurations. Security measures like API key authentication, HTTPS support, and IP whitelisting will be implemented, with thorough documentation provided on installation, configuration, and API usage. With expertise in Linux, SIP, and REST APIs, I am well-equipped to deliver a robust, scalable solution that meets your requirements.

@OutsourceMan

With our experience of nearly two decades, CnELIndia is equipped with the skills and proficiency needed to handle your FreePBX / Asterisk project. Our notable successes in web and app development make us capable of creating a tailored solution that meets all your needs. From installation and configuration to REST API development, we've got you covered. Our knowledge of Node.js, PHP, and Python will prove instrumental in integrating a thoroughly functional REST API for controlling the PBX. We will ensure that every call state is meticulously tracked and clear status and error codes are returned for easy understanding. Moreover, our expertise in managing large-scale systems and data handling uniquely qualifies us for the task of recording each call and transferring them via robust methods to your remote server. Most critically, we understand the importance of security in telephony systems. With API Key authentication, HTTPS support, IP whitelist protection for every API Key, along with permission validation on every request, activity logs, access logs, and error logs, we guarantee a system that is secure against any unauthorized access. Beyond this project's needs, our proven capacity for designing multi-project supported systems will enable easy scalability while maintaining complete data isolation. Let us demonstrate our skills by making your telephone bridge calls fully automated yet thoroughly reliable!

@diektech

Hi — Elias here from Miami. I see you're looking for a FreePBX/Asterisk installation and configuration, which is crucial for seamless communication. Your goal seems to be establishing a reliable telephony system that integrates well with your existing workflows. The real technical challenges often lie in scalability and reliability, particularly with complex SIP configurations. A common issue in setups like this is managing integrations and ensuring efficient call flow logic. It looks straightforward initially, but nuances in permissions and state management can lead to complications. My approach would involve assessing your current infrastructure and using Docker for a consistent installation environment. I'd prioritize clear documentation for maintenance and future updates, making it easier for your team to manage the system. I've worked on several telephony systems and integrations, focusing on performance and minimizing downtimes. A few questions to better understand the scope: Q1 – What specific user roles and permissions do you envision for the FreePBX system? Q2 – Are there existing applications that need to integrate with this setup? Q3 – What are your expectations for scalability as your user base grows? Happy to go through the details and suggest the best technical approach. Looking forward to hearing from you.

@a4logic

With over a decade of experience, my team at Our Software has become adept at working with web services, and we specialize in complex integrations. This makes us the perfect fit for your FreePBX / Asterisk project. Our past projects have included developing and designing websites using the latest technologies and creating Content Management Systems, skills that will come in handy as we configure your PBX system and develop a REST API for you. We understand the importance of a redundant, secure, and documented system, so these will be our focus when setting up your SIP trunks and automating call bridging through the API. In addition to this, we have extensive experience in developing robust record management systems, such as what you're looking for with your call recordings. We can ensure your messages are securely saved on a remote server using reliable transfer methods like SCP or SFTP while easily retrievable through our API.

@kamran2012

Hey, I will install FreePBX, configure your SIP trunks with multiple DDI support, and build the full REST API for bridge calls with real time webhook events and multi-project isolation. On a similar telephony build, using ARI for the bridge call flow (instead of AMI originate) gave much more reliable call state tracking across all the stages you listed. I will wire yours the same way. Questions: 1) For the remote recording storage, is SFTP already set up on that server, or do you need me to configure it as well? 2) Do you have a preferred framework for the REST API (Python/Flask, Node, PHP), or is that open? Share the SSH access and SIP trunk details and I will have Phase 1 running on your server within a few days. Looking forward to talking through the details. Kamran

@techplusintl

Hi, To automate telephone bridge calls, I will install and configure FreePBX on your Linux server, ensuring it supports multiple SIP trunks and automatic call recording. I will then develop a REST API that allows for seamless bridge calls, complete with detailed call state reporting and webhook notifications. I have extensive experience with FreePBX and Asterisk, and I prioritize clean, maintainable code. I am also open to suggestions for improving the architecture. Happy to discuss the details.

@KatrinaQ

Hi, I’m a senior telecom developer with 20+ years of experience delivering production-grade Asterisk/FreePBX systems and REST APIs, including similar bridge-call automation projects, so this is straightforward. I’ll install FreePBX on your Linux server, configure SIP trunks for multiple DDI support, enable automatic call recording, and harden the setup for production. Next, I’ll build a secure REST API using FreePBX ARI and AMI that handles your exact call-flow, real-time webhooks, whitelist management, and remote SCP recording uploads, isolating each project’s configuration, logs, and API keys. I’ll include call-status queries, cancellation, and audited history retrieval, then provide Swagger docs and a deployment checklist. I can start today after a quick server review.

@starttechno

For the bridge flow you described — call A, wait for answer, then dial B while A hears ringback, and connect both once B picks up — the reliable way to build this is on Asterisk's ARI (Stasis), not the FreePBX dialplan, since ARI gives channel-level control over originate, ringback and bridging. I'd originate to A, create a mixing bridge on answer, play ringback to A while originating B, then drop B in when B answers. Every SIP cause maps cleanly to your states — A busy, B did not answer, Bridge established and the rest — each pushed to your webhook with a unique Call ID for status queries. For install I'll set up FreePBX, add your SIP trunks with multi-DDI so each request picks the trunk and Caller ID to show, run several trunks at once, enable MixMonitor recording shipped to your remote store, and harden the box with fail2ban, firewall and TLS/SRTP. The API itself I'd build in Node.js or FastAPI over ARI. One thing on A hearing the ringing tone while B rings: do you want the real early-media ringback from your SIP provider passed through, or a locally generated tone? That changes the trunk config. Muhammad Saad

@seefattechnologi

I got the scope and am pretty much good to get started on this. I can deliver a production-ready FreePBX/Asterisk solution with a secure REST API, bridge-call automation, ARI/AMI integration, webhooks, multi-project isolation, call recordings, and complete Swagger/Postman documentation using clean, extensible architecture. Please ping me to get started and provide you outstanding results. Thanks!!

@ravichedwal

With over 8 years of experience in Full Stack Development focused primarily on PHP and Python, I am confident that my skills and expertise align perfectly with your project needs. Having worked on a multitude of projects involving API integrations, SIP trunks, webhooks, and secure backend development, I have amassed a deep understanding of the intricacies involved in managing large-scale telephony systems like the one you describe. What sets me apart is not just my technical know-how but also my dedication to client satisfaction. I ensure a continuous feedback loop with my clients through daily and weekly reports, fostering open communication and ensuring that your vision for the project is executed to perfection. My coding practices prioritize scalability, stability, and robust security measures -all key components required for your production-ready telephone bridge calls system. Finally, although I'm comfortable working independently, I recognize the importance of working collaboratively as well. I have no doubt that we would make an incredible team, bringing out the best of each other's ideas and expertise to create the finest possible solution for your business. Let's connect to explore how I can contribute value to your project and help you achieve the operational efficiency you're looking for!

@godesignpk

Good afternoon, I can help install and configure FreePBX with multiple SIP trunks and develop a secure, production-ready REST API for your call bridging system. I will focus on SIP, API development, and Linux server setup to ensure stable call handling, recording, and webhook event notifications. I will deliver clean, maintainable code with full documentation and secure API access including IP whitelisting and activity logs. Do you already have the server access and SIP trunk details ready for the initial setup? Best regards, Waqas & GoDesign Team

@intelliresponse

Hi! I have extensive experience with FreePBX/Asterisk, SIP, and REST API development. I can build a secure, production-ready PBX with bridge calling, webhooks, call recordings, multi-project isolation, and complete API documentation. Let's discuss your requirements and get started.

@devendrathakur12

Hello, I recently worked on a PBX automation project where I built custom call routing APIs and automated call handling with detailed event tracking. I can build a clean FreePBX/Asterisk solution with bridge calls, webhooks, recordings, multi-project isolation, and secure REST APIs that is easy to maintain and extend. One question: do you prefer ARI or AMI for the bridge call flow, or are you open to whichever provides better reliability and scalability? Let's build something your future developers will thank us for, not complain about. ? Dev S.

@ozbas

As an experienced Full-Stack software engineer for over a decade, my expertise aligns perfectly with your FreePBX/Asterisk project. I am well-versed in working with APIs and have developed numerous REST APIs that have been commercially successful. My knowledge of Node.js will be especially valuable for developing the PBX controlling API you need, enabling me to deliver exactly what you're looking for, including detailed call handling states and a webhook-based event system. Apart from successfully configuring FreePBX or Asterisk, I excel at securing PBX systems and ensuring their optimal performance. Producing a production-ready, stable, secure, and well-documented solution backs up my claims. Moreover, my extensive experience in multi-project support with isolated configurations guarantees your data will be strictly separated while allowing seamless management of essential aspects like API Key authentication, HTTPS support, IP whitelist protection, etc. All projects will have their own isolated configuration as required including API key management, SIP trunks as well as the allowed caller IDs & IP addresses reinforcing the security layers even further. I'll provide you protection against unauthorized access complemented by thorough logs tracking right from error logs to access/logs to activity logs.

@b2btechnology20

As an experienced developer who has extensively worked on FreePBX and Asterisk, I am a perfect fit for this project. I understand the importance of setting up a stable, secure and reliable PBX system as well as the criticality of automating telephone bridge calls for smooth communication operations. In my over 11+ years of experience in solving complex web development issues, I have consistently delivered high-quality, well-documented solutions to clients. Lastly, I ensure a secure and robust communication system as per your needs through in-built systems like an API Key authentication, HTTPS support, IP whitelist protection among others are never compromised. My endeavor throughout this intricate task would be to deliver not just a technically-enabled solution but also a user-friendly system which could serve multiple independent projects cleanly. Let's collaborate to bring this vision into reality!

@stelianrosca618

hello, I have 5+years experience with Python, PHP and Docker And then I am adopted on Linux environment So i can start work immediately

@umairdev101

I am an expert VoIP engineer with deep experience in FreePBX and Asterisk deployments across cloud and on-premise Linux environments. I can efficiently handle the FreePBX Install and Call API project by configuring a secure, stable PBX instance and developing a robust REST API for automated call initiation. I have previously built custom PHP and Node. js middleware to bridge Asterisk AMI/ARI with external web applications, ensuring seamless call control and real-time logging. My workflow includes implementing Docker for scalable deployments and fine-tuning SIP settings for optimal audio quality. I am ready to start immediately and can provide a streamlined interface for your calling requirements. Please message me to discuss your specific API endpoints and server environment.

@myGigs

Good morning. I'll install and configure FreePBX with your SIP trunks, multi-DDI support, and automatic call recording, then build a production-ready REST API in PHP or Node.js covering the full bridge call flow, all call state reporting, webhook events, whitelist management endpoints, and automatic recording uploads to your remote server via SFTP or SCP. Looking forward to hearing from you.

@vishal1145

Your bridge call system will fail under concurrent load if you do not implement proper channel locking and SIP session management from day one. Without queue-based call orchestration, you will hit race conditions when multiple API requests try to originate calls simultaneously. Quick questions - what is your expected peak concurrent call volume, and are you planning active-active failover or just single-node deployment with backups? And do your SIP trunks support SRV failover or will you need application-layer retry logic? Here is the architectural approach: - FREEPBX + ASTERISK AMI: Deploy containerized FreePBX with custom AMI event handlers that push real-time call state changes to your webhook endpoints via Redis pub/sub for sub-100ms latency. - REST API (NODE.JS): Build FastAPI-based control plane with JWT-scoped API keys, per-project channel pools, and PostgreSQL for whitelist/recording metadata with row-level security enforcing tenant isolation. - CALL RECORDING PIPELINE: Implement post-call async workers that compress recordings to Opus format and push to your SFTP endpoint with retry queues, reducing storage costs by 70% compared to raw WAV files. I have built similar telephony platforms for 2 healthcare companies processing 10K+ daily bridge calls with HIPAA-compliant recording retention. Let's schedule a 20-minute technical call to validate your SIP provider's API limits before we finalize the architecture.