FusionPBX / FreeSWITCH & SIP Telephony Engineer — Part-Time

FusionPBX / FreeSWITCH & SIP Telephony Engineer — Part-Time Remote | 10–20 Hours/Week | Hourly About SkipHold SkipHold is an AI communications platform that connects conversational AI with existing PBX and telecom infrastructure. We work with MSPs, UCaaS providers, telecom providers, and businesses to integrate AI voice agents with traditional phone systems without requiring them to replace their existing telephony infrastructure. About the Role We're looking for an experienced VoIP / SIP engineer to provide ongoing part-time support for our telephony infrastructure. This is a hands-on technical role focused primarily on FusionPBX, FreeSWITCH, SIP infrastructure, PBX integrations, and real-world call-flow troubleshooting. You'll work alongside our software and telephony engineering team to configure and validate PBX environments, troubleshoot production issues, test automated provisioning, and ensure our AI ↔ PBX integrations work reliably end-to-end. What You'll Work On Configure, troubleshoot, and validate FusionPBX / FreeSWITCH environments Create and validate PBX tenants/domains, extensions, SIP credentials, gateways, DIDs, and dial plans Configure and troubleshoot SIP trunks and inbound/outbound call routing Configure and troubleshoot ring groups, queues, IVRs, and fallback/no-answer routing Test and troubleshoot blind, attended, and warm transfers, SIP REFER, and AI-to-human handoffs Troubleshoot SIP signaling, RTP, NAT, codecs, registrations, audio, and call-routing issues Validate AI ↔ PBX call flows and connections between AI agents and PBX extensions Test newly automated PBX provisioning and configuration changes Troubleshoot carrier, PBX, softphone, and endpoint interoperability Work with developers to determine whether issues originate in the application, PBX, SIP configuration, carrier, network, or endpoint Document repeatable configurations and troubleshooting procedures Required Experience Strong hands-on experience with: -FusionPBX -FreeSWITCH -SIP / RTP / VoIP -SIP trunks, gateways, DIDs, extensions, and dial plans Linux administration -NAT, firewalls, DNS, TLS/SRTP, and networking -SIP troubleshooting using tools such as sngrep, Wireshark, tcpdump, and FreeSWITCH CLI -Diagnosing complex call-routing, registration, and audio issues end-to-end We're looking for someone who can independently trace a call flow, identify where it's failing, and work with our engineering team to resolve the underlying issue. Nice to Have Experience with any of the following is a plus: NetSapiens FreePBX / Asterisk 3CX Ringotel Twilio / BYOC OpenSIPS / Kamailio DigitalOcean / AWS REST APIs / Postman PostgreSQL AI voice platforms such as Vapi, LiveKit, Retell AI, OpenAI Realtime, or ElevenLabs AI experience is helpful but not required. Strong PBX, FreeSWITCH, SIP, and telephony troubleshooting expertise is considerably more important. Availability This is intended to be an ongoing part-time relationship of approximately 10–20 hours per week. We value someone who can maintain some flexibility for scheduled deployment and validation sessions, as well as occasional time-sensitive troubleshooting, where our software and telephony engineers may need to test and iterate on changes together in real time. When Applying Please include a short description of your hands-on experience with FusionPBX, FreeSWITCH, and SIP troubleshooting, including the types of PBX environments you've supported. Candidates who appear to be a strong fit may be asked to complete a short technical assessment based on realistic PBX/SIP troubleshooting scenarios before moving to an interview.