FreePBX/Twilio Outbound Call Issue Resolution

@farooqtahir2013

As an experienced Network, VoIP, and Systems Engineer with a decade-long background in troubleshooting complex IT issues, I have the right set of skills that aligns perfectly with your project requirements. I’ve not only integrated Twilio Elastic SIP Trunks with FreePBX servers numerous times but also ensured smooth communication by thoroughly analyzing every element including PJSIP trunk settings, outbound routes, and SIP signaling. With my deep understanding of Asterisk CLI and other professional tools like sngrep and tcpdump, I can effortlessly trace the fault and rectify it within no time. What differentiates me from other bidders is my approach - I don’t just resolve the issue at hand; instead, I opt for a detailed analysis while providing all the necessary adjustments to reproduce on other servers if needed. Being well-versed with FreePBX, PJSIP trunks, and Twilio Elastic SIPs makes me competent enough to go beyond a narrow troubleshooting scope. Moreover, my core proficiencies in Network administration (Routing, Switching, VPNs), Linux servers administration (essential for managing asterisk), along with my unbeatable record of 100% project completion makes me your ideal choice. I assure you that once I'm done with the task at hand, you'll experience two-way audio calls without any annoying "All circuits are busy now" error message and those successful calls will promptly reflect on Twilio's live call logs as expected. Let's make this project shine!

@topworkerbd

Hi , I've spent years architecting and hardening Asterisk/FreePBX-based telecom platforms — IVR, OBD, and enterprise voice systems — including live PJSIP trunk integrations with Twilio Elastic SIP Trunks. I've resolved this exact "all circuits busy with zero Twilio log entries" pattern before; it's almost always Asterisk rejecting the call locally before it ever reaches Twilio — usually a mismatch in the endpoint's identify/match, from_domain, or contact settings versus what Twilio's Origination expects, or an outbound route pattern silently mangling the dialed number. My approach: Capture live signaling with sngrep/tcpdump plus pjsip set logger on during a test call to see exactly where the INVITE fails or gets rejected. Audit the PJSIP endpoint/aor/identify/transport config against Twilio's required parameters (from_domain, NAT settings, codecs). Correct the trunk/outbound route mismatch, then re-test until we get two-way audio and a confirmed Twilio call log entry. Hand you a short, reproducible runbook of the exact fix for your other servers. I can start immediately and typically resolve trunk-level SIP issues like this within a single focused session.

@rachitphuyal

I have experience troubleshooting FreePBX, Asterisk, PJSIP, and Twilio Elastic SIP Trunks. I can trace the issue using the Asterisk CLI, sngrep/tcpdump, and verify the outbound route, PJSIP trunk, authentication, and SIP signaling until calls successfully reach Twilio with two-way audio. I'll also provide a concise summary of the changes made. Ready to start immediately.

@shaxzodj

Hello BosS A quick question: Are you using Twilio's Credential List authentication or IP Authentication for the Elastic SIP Trunk?

@tatumm17

I can help you trace and fix the “All circuits are busy” issue by walking through your outbound route logic, PJSIP trunk, and SIP signaling so that calls actually hit Twilio and complete with audio. I’ve worked extensively with FreePBX, PJSIP trunks, and Twilio Elastic SIP, including diagnosing one-way audio, codec mismatches, TLS/SRTP, and 403/488 issues that never show in Twilio logs when signaling fails upstream. My approach is to reproduce the error, capture with sngrep/tcpdump and Asterisk CLI, correct the trunk and route settings, then verify successful test calls and document each change for reuse on your other servers. I would love to chat more about your project! Regards

@HafizMArhamAli

Hi, I have solid experience in this type of work and can handle exactly what you’re looking for with clean, professional results. I focus on delivering quality, staying on schedule, and making sure everything meets your expectations. I’m ready to get started and can adapt to your preferred style or requirements. We can discuss the details further in chat to clearly define the scope and make sure everything is aligned. Looking forward to working with you. Thanks,

@mohtashimjaved12

Dear hiring manager, Deliver: 24 hours Siptrunk: available for all over world Video: attached I've read your proposal carefully, and this is exactly the kind of problem we solve. We've already built and run our own production PBX/call-center platform on Asterisk, with full custom development on top of it, so we're not coming in to "learn ViciDial" or stitch together off-the-shelf modules — we're coming in with a working system and the ability to extend it to solve your specific routing problem. Our current platform already includes: ✅ Auto Dialing System ✅ Bulk Lead Upload (CSV) ✅ Smart IVR Routing ✅ Live Call Transfer ✅ Voicemail Drop ✅ Local Caller ID Display ✅ Real-Time Reporting ✅ CRM Integration ✅ 24/7 Support ✅ Agents Transfer calls One system. Every outbound need covered