VoIP Engineer

Position: VoIP Engineer – Asterisk / FreePBX Department: Technology / Infrastructure Employment Type: Full-Time Experience: 3+ Years Job Summary We are looking for an experienced VoIP Engineer with strong hands-on expertise in Asterisk, FreePBX, SIP, RTP, WebRTC and IP networking. The engineer will be responsible for managing our VoIP infrastructure, SIP trunks, CRM calling integration, extensions, inbound/outbound routing and troubleshooting production call issues including one-way/no audio, call drops, registration failures, NAT/firewall problems and SIP/RTP connectivity issues. The candidate should be comfortable diagnosing complex issues using SIP traces, Asterisk logs, packet captures and networking tools. Key Responsibilities Install, configure, maintain and troubleshoot Asterisk and FreePBX servers. Configure and manage PJSIP/SIP extensions, trunks and endpoints. Configure inbound routes, outbound routes, dial plans, IVRs, queues, ring groups and call recording. Integrate FreePBX/Asterisk with CRM and browser-based calling systems. Configure and troubleshoot WebRTC, WebSocket (WS/WSS), SIP over TLS and RTP. Troubleshoot: One-way audio No audio Call drops SIP registration failures 401/403/404/408/480/486/503 SIP errors RTP/media problems NAT traversal issues Codec negotiation problems SIP trunk failures Inbound/outbound routing problems Analyze SIP signaling and RTP traffic using sngrep, Wireshark, tcpdump and Asterisk CLI/logs. Configure and troubleshoot SIP trunks with telecom/VoIP providers. Work with providers to diagnose routing, DID, CLI/caller-ID and connectivity issues. Configure NAT, firewall rules, port forwarding, VPN and network routing for VoIP infrastructure. Monitor latency, jitter, packet loss and overall voice quality. Configure QoS where required to improve voice traffic performance. Maintain secure VoIP infrastructure and prevent unauthorized SIP access and toll fraud. Perform FreePBX/Asterisk upgrades, backups and recovery. 1 Monitor production telephony infrastructure and respond to critical incidents. Maintain technical documentation for trunks, extensions, dial plans, firewall rules and call flows. Coordinate with CRM/backend/networking teams for new integrations and troubleshooting. Required Technical Skills Must Have Strong hands-on experience with Asterisk & FreePBX Strong understanding of SIP and RTP PJSIP configuration and troubleshooting SIP trunk configuration Inbound/outbound routing Asterisk dial plans IVR, queues and ring groups WebRTC / WebSocket / WSS NAT traversal TCP/IP, UDP and DNS Linux administration Firewall and networking knowledge Wireshark tcpdump sngrep Asterisk CLI/debugging SIP/RTP packet analysis Networking Knowledge Candidate should understand: TCP/UDP NAT/PAT Public/private IP addressing DNS VLANs VPN Firewall configuration RTP port ranges QoS TLS/SSL STUN/TURN/ICE concepts Network latency, jitter and packet loss Preferred Skills Experience with any of the following will be an advantage: Session Border Controllers (SBC) Kamailio / OpenSIPS FreeSWITCH 2 3CX AWS or other cloud infrastructure Linux – Ubuntu/Debian/CentOS Nginx SSL/TLS certificates Python/Bash scripting REST APIs WebSockets CRM/Contact Center integrations High-availability VoIP architecture VoIP monitoring and alerting Experience & Qualification 3+ years of hands-on VoIP engineering experience Strong production experience with Asterisk/FreePBX Previous experience managing SIP trunks and production calling infrastructure Experience supporting call center/CRM telephony environments is strongly preferred Bachelor's degree in Computer Science, IT, Telecommunications, Electronics or related field preferred but not mandatory for candidates with strong practical experience. Ideal Candidate We are specifically looking for someone who can independently diagnose a production call issue from end to end: CRM/Browser → WebRTC/WSS → Asterisk/FreePBX → SIP Trunk → Telecom Provider The candidate should be able to inspect SIP signaling and RTP traffic and quickly determine whether an issue originates from the browser/CRM, PBX, firewall/NAT, network, RTP configuration, SIP trunk or telecom provider

Pay: AED5,664.17 - AED8,000.00 per month

Work Location: In person