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.
- 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
- 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
Preferred candidate profile
- 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)
- 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.
- Kamailio/OpenSIPS or SBC exposure
- AMI/ARI or ESL • WebRTC, TLS/SRTP • kernel tuning • any cloud platform, Python, or monitoring tool
