Cloud Engineer (Telephony)

About the Role

About the role

You will own and manage our telephony platform and the Linux systems. The cloud and DevOps piece is deliberate our platform runs on cloud infrastructure, and we want the person maintaining it to eventually deploy and automate it themselves. One doesnt need Cloud & DevOps

experience

coming in. We hire on telephony and Linux fundamentals; you'll have the mentorship, the time, and the real ownership to build the rest here. Role &

responsibilities

Telephony - Build and maintain Asterisk and FreeSWITCH installation, upgrades, configuration, hardening - Write and debug dial plans: routing, IVR, queues, time conditions, failover - Root cause from SIP traces and RTP streams Linux (equally core) - Own the servers: provisioning, patching, hardening, capacity, backups, recovery - Diagnose system-level performance problems - Manage the networking stack: routing, DNS, NTP, firewalls, and the packet path a call actually takes Cloud & DevOps (where the role grows ) - Cloud fundamentals, then automation in Bash or Python, then monitoring, containers, and CI/CD as you get comfortable

Requirements

Preferred

candidate profile

Telephony - 2 to 3 years hands-on with Asterisk and/or FreeSWITCH in production - Real dial plan authoring and debugging, not just editing someone else's -

Experience

with SIP trunk providers and PSTN connectivity Linux (good to expert level) - Command-line fluency - Systematic troubleshooting on an unfamiliar box under load - Networking knowledge: TCP/UDP behaviour, routing, DNS, NAT, firewalls, and how to prove where a packet is dropped Troubleshooting (What matters more than the checklist) - You want to know why it broke and get to the root cause of the issue to fix it. You reach for the layer below when the obvious answer doesn't hold. Good to have - Kamailio/OpenSIPS or SBC exposure - AMI/ARI or ESL

• with SIP trunk providers and PSTN connectivity Linux (good to expert level) - Command-line fluency - Systematic troubleshooting on an unfamiliar box under load - Networking knowledge: TCP/UDP behaviour, routing, DNS, NAT, firewalls, and how to prove where a packet is dropped Troubleshooting (What matters more than the checklist) - You want to know why it broke and get to the root cause of the issue to fix it. You reach for the layer below when the obvious answer doesn't hold. Good to have - Kamailio/OpenSIPS or SBC exposure - AMI/ARI or ESL
• WebRTC, TLS/SRTP
• kernel tuning
• any cloud platform, Python, or monitoring tool

Preferred

candidate profile

Telephony - 2 to 3 years hands-on with Asterisk and/or FreeSWITCH in production - Real dial plan authoring and debugging, not just editing someone else's -

Experience

with SIP trunk providers and PSTN connectivity Linux (good to expert level) - Command-line fluency - Systematic troubleshooting on an unfamiliar box under load - Networking knowledge: TCP/UDP behaviour, routing, DNS, NAT, firewalls, and how to prove where a packet is dropped Troubleshooting (What matters more than the checklist) - You want to know why it broke and get to the root cause of the issue to fix it. You reach for the layer below when the obvious answer doesn't hold. Good to have - Kamailio/OpenSIPS or SBC exposure - AMI/ARI or ESL

• with SIP trunk providers and PSTN connectivity Linux (good to expert level) - Command-line fluency - Systematic troubleshooting on an unfamiliar box under load - Networking knowledge: TCP/UDP behaviour, routing, DNS, NAT, firewalls, and how to prove where a packet is dropped Troubleshooting (What matters more than the checklist) - You want to know why it broke and get to the root cause of the issue to fix it. You reach for the layer below when the obvious answer doesn't hold. Good to have - Kamailio/OpenSIPS or SBC exposure - AMI/ARI or ESL
• WebRTC, TLS/SRTP
• kernel tuning
• any cloud platform, Python, or monitoring tool