Fix FreePBX-Twilio Outbound Calls

@sanarena

Hi, I can fix your FreePBX outbound calls so they reach Twilio reliably. I’ll start by verifying the PJSIP trunk’s match rules and outbound route dial patterns, then run a SIP trace from the Asterisk CLI to confirm whether the INVITE is being sent to the correct Twilio endpoint. If the trace shows no attempt, I’ll check the trunk’s context and caller ID settings against Twilio’s requirements. I’ll also review the credential list binding and ensure the From header matches a verified Twilio number or your trunk’s origination URI. Once corrected, I’ll place test calls and confirm they appear in your Twilio logs. Could you share the current PJSIP trunk configuration and the outbound route details? I can deliver this within 20 days for INR 1250.

@kishanbwts

I’ve encountered similar issues where outbound calls from FreePBX to Twilio show “All circuits are busy” but the SIP endpoints seem available. Usually, this is due to a PJSIP trunk misconfiguration or a missing/incorrect route that blocks SIP INVITEs before reaching Twilio. My approach starts with detailed tracing using sngrep and Asterisk CLI to verify if the INVITE leaves the server and how the SIP signaling behaves. I’ll check the credential list bindings, outbound route dial patterns, and PJSIP trunk settings like endpoint identification and transport. Then I’ll apply corrections directly in FreePBX and test live calls for audio and proper logging at Twilio. Do you currently have SIP OPTIONS ping enabled on the PJSIP trunk? Sometimes disabling or tweaking this helps signaling sync. Also, is your outbound route explicitly routing calls through the Twilio trunk or relying on a catch-all route? I’ll fix the problem end-to-end and provide a straightforward summary of the configuration updates you can apply elsewhere. Ready to dive in and resolve this quickly.

@iimtpradeepdash

When outbound calls from FreePBX hit “All circuits are busy” but Twilio call logs are empty, the SIP traffic isn’t leaving your box—most often the trunk config or outbound route is off, or FreePBX isn’t authenticating correctly with Twilio’s SIP spec. I’ve fixed FreePBX-Twilio setups like this several times, both with IP auth and Credential Lists. Last one was a law firm PBX running two trunks, PJSIP, confirmed with sngrep and tcpdump. I’d run live call attempts, trace SIP packets, check Contact/User/From headers, then correct any signaling or route issue in FreePBX. After fixing, I’ll show you a completed call in Twilio logs, with working audio, and list changes made for your records. Does your FreePBX sit behind NAT? That changes a few trunk settings. Pradeep

@DigitalNightOwl8

Hello Mate!Greetings , Good afternoon! I’ve carefully checked your requirements and really interested in this job. I’m full stack node.js developer working at large-scale apps as a lead developer with U.S. and European teams. I’m offering best quality and highest performance at lowest price. I can complete your project on time and your will experience great satisfaction with me. I’m well versed in React/Redux, Angular JS, Node JS, Ruby on Rails, html/css as well as javascript and jquery. I have rich experienced in Twilio, Network Monitoring, VoIP, Asterisk PBX, Linux, SIP and PHP. For more information about me, please refer to my portfolios. I’m ready to discuss your project and start immediately. Looking forward to hearing you back and discussing all details.. Thanks

@sheikhyaseen9249

Hello, I hope you're doing well. I understand you're experiencing outbound call failures on your FreePBX server. Since Twilio isn't receiving any SIP traffic and you're getting "All circuits are busy now," the issue is likely within the FreePBX outbound route, PJSIP trunk configuration, or SIP signaling before the request reaches Twilio. I can systematically troubleshoot the problem using the Asterisk CLI, PJSIP diagnostics, sngrep/tcpdump, and FreePBX logs to isolate the root cause. I'll verify the outbound routes, trunk settings, authentication, dial patterns, codecs, SIP headers, and signaling, then apply the necessary fixes until outbound calls complete successfully with two-way audio and appear in Twilio's call logs. Once everything is working, I'll provide a concise summary of the changes made so you can replicate the configuration on your other servers. ✔️ Experience troubleshooting FreePBX & Asterisk ✔️ Strong knowledge of PJSIP trunk configuration ✔️ SIP debugging using Asterisk CLI, sngrep & tcpdump ✔️ Fast communication and prompt resolution I'm available to start immediately and would be happy to help resolve this issue. Best regards, Muhammad Yaseen

@irfanui

Hi, I reviewed your project: Fix FreePBX-Twilio Outbound Calls. I can help you build, fix, or upgrade your Laravel/PHP application with secure authentication, database design, REST APIs, admin panels, payment integration, optimization, and deployment. I have strong experience with production Laravel systems, MySQL, JavaScript/React, API integrations, and server deployment. Please message me with the current codebase status and feature list so I can suggest the safest implementation plan. Portfolio: https://www.freelancer.com/u/irfanui Regards, Mohammad 4th Dimension Partners

@kuldeepsharma30

Hello, I have experience troubleshooting **FreePBX, Asterisk/PJSIP, and Twilio Elastic SIP Trunks**, including outbound routing, SIP signaling, authentication, and media negotiation issues. For this issue, my approach will be: * Review the **Outbound Route**, Trunk configuration, and Dial Patterns in FreePBX. * Verify the **PJSIP trunk** (Authentication, AOR, Endpoint, Identify, Transport, and Contact settings). * Use **Asterisk CLI**, **pjsip set logger on**, **sngrep**, and **tcpdump** to trace the SIP signaling and identify why calls never reach Twilio. * Check Twilio Elastic SIP settings, Credential List authentication, Origination URI, IP ACLs (if applicable), codecs, caller ID formatting, and E.164 dialing requirements. * Apply the necessary corrections and perform end-to-end testing until outbound calls complete successfully with two-way audio. **Deliverables:** * Fully working outbound calling through Twilio. * Verification that calls appear in the Twilio Call Logs. * A concise document outlining the root cause, configuration changes, and recommendations for replicating the setup on other servers. I'm available to start immediately and can troubleshoot the issue remotely via SSH or your preferred remote access method. Best regards, **Naveen**

@Jennycreation

With over 17 years of comprehensive experience, I am confident that I can effectively resolve the issues you are experiencing with your FreePBX and Twilio setup. My expertise in PHP and Twilio makes me well-equipped to handle the intricacies of your project concerning PJSIP trunks, Twilio Elastic SIP, and FreePBX troubleshooting. My first step towards resolving this problem would be performing a meticulous trace and isolating the fault across the outbound route, PJSIP trunk, and SIP signaling to pinpoint the issue accurately. Once I've identified the root cause, I will make the necessary corrections directly in FreePBX ensuring that calls not only reach Twilio but complete successfully with two-way audio as your Acceptance Criteria specifies. One aspect that sets me apart is my commitment to knowledge transfer. At the end of the project, I will provide you with a concise summary of all adjustments made. This will enable you to replicate these adjustments on other servers if needed. Having delivered high-quality projects throughout my career, I am confident in my ability to meet your project expectations while providing exceptional value for your investment. Let's make those "busy circuits" a thing of the past together!

@ravikumarsingh1

I have completed similar VoIP, Linux server administration, SIP trunk, and production troubleshooting projects. I also manage Linux production servers and have extensive experience debugging networking, firewalls, Nginx, Docker, and real-time services. If we start working together, I'd be happy to show you similar server troubleshooting work. I have experience troubleshooting FreePBX, Asterisk, PJSIP, SIP trunks, Twilio, Linux, and network-level issues. From your description, since Twilio shows no inbound INVITEs, the issue is likely before the traffic reaches Twilio—typically involving the outbound route, PJSIP trunk configuration, authentication, DNS, NAT, or SIP signaling. I can help by: • Analyze the outbound route and trunk configuration • Debug SIP signaling using Asterisk CLI, sngrep, and tcpdump • Verify PJSIP endpoint, authentication, codecs, NAT, and transport settings • Review Twilio Elastic SIP Trunk configuration and Credential List authentication • Apply the required fixes and validate successful outbound calling • Verify two-way audio and confirm calls appear in the Twilio Call Logs Deliverables: ✔ Fully working outbound calls through Twilio Elastic SIP Trunk ✔ Two-way audio verification ✔ Calls visible in Twilio Call Logs ✔ Resolution of the "All circuits are busy now" issue ✔ Configuration summary and documentation I'm available to start immediately and can troubleshoot the issue over SSH or screen sharing until outbound calling is fully operational.

@heenafullstacken

Your Laravel requirement for Fix FreePBX-Twilio Outbound Calls fits my production-development experience. I can build new modules, resolve complex issues, or safely upgrade the application with authentication, MySQL, REST APIs, admin panels, payments, optimization, and deployment. Please share the Laravel version, current code status, and priority feature or problem. I will review it and recommend the safest path forward. Relevant work: https://www.freelancer.in/u/heenafullstacken Regards, Heena | A Plus IT House

@tatumm17

I can pinpoint and fix why your FreePBX PJSIP trunk to Twilio never reaches their side, then verify outbound calls work without “All circuits are busy” and show correctly in Twilio’s logs. I’ve configured and troubleshot multiple FreePBX/Asterisk systems with PJSIP trunks, Twilio Elastic SIP, and Zoiper endpoints, including signaling issues, codec mismatches, and authentication problems. I’d trace the call path with sngrep and Asterisk CLI, correct trunk and route settings, then document the exact changes so you can replicate them on other servers. I would love to chat more about your project! Regards

@infovergetech

We have experience in troubleshooting FreePBX and Twilio SIP integrations, resolving outbound calling issues, configuring SIP trunks, dial plans, authentication, codecs, and ensuring reliable call routing with detailed log analysis. Our company has 17+ years of experience in IT service development. Couple of quick questions: Are inbound calls working correctly, and is the issue only with outbound calls, or are both directions affected? Would you like us to perform a complete SIP trunk health check, including firewall, NAT, and Twilio configuration, to prevent similar issues in the future? You might see our profile is new, but not new in this business. Kindly open the chatroom, we even provide you some free additional development if required. Also release the payment once we finish the task as you prefer. Give us an opportunity and we won't fail you. Thanks, Sandeep K.

@MuhammadAkif8

With my extensive experience as a Full Stack developer and a special focus on PHP-based frameworks such as FreePBX, I'm confident that I can solve your Twilio outbound call issue. I have worked on numerous projects that include configuring PJSIP trunks and utilizing Twilio Elastic SIP in an innovative manner. One particular project stands out where I successfully traced and fixed a similar problem within the routing system of a client's FreePBX server. Not only will I remedy the current issue, but I'm also committed to providing you with a clear report of the adjustments made, enabling you to replicate the same for other servers. My knack for simplifying complex problems into manageable solutions will greatly facilitate your future troubleshooting tasks. Additionally, my global work experience has sharpened my adaptability and honed my project management skills. You can rely on me for responsive and transparent communication throughout our collaboration. Let's leverage my skills and expertise to ensure successful, two-way calls while leaving no traces of "All circuits are busy" messages. Best, Akif F