VoIP Engineer

Описание вакансии

We’re Hiring: VoIP Engineer
📍 Location: Remote
👤 Reports to: CTO
💡 About Us
Yesim is an international IT company that develops and promotes digital eSIM products for the global market.
We help travelers, digital nomads, and business professionals stay connected anywhere in the world — without physical SIM cards.
Today, more than 2.5 million users across Europe, the Americas, Asia, and the Middle East trust Yesim.
We partner with 800+ mobile operators, maintain a 4.5★ Trustpilot rating, and have over 27,000 user reviews.
About the Role
We are looking for a VoIP Engineer responsible for developing, maintaining, and scaling our SIP-telephony infrastructure.
Your mission is to ensure high-quality voice services under heavy load, resolve incidents efficiently, and implement improvements that directly influence product quality and stability.
🛠 Responsibilities
  • Design, configure, and maintain VoIP infrastructure (Asterisk, FreeSWITCH).
  • Administer and optimize SIP traffic, call routing, and dialplans.
  • Integrate VoIP platforms with CRM/ERP/internal systems (API, webhooks, AMI/ARI).
  • Configure and support SBCs, SIP proxies, and RTP media servers.
  • Diagnose and resolve incidents using traffic debugging, log analysis, and pcap review.
  • Connect and evaluate new VoIP providers, ensuring high voice quality.
  • Configure IVR, queues, load distribution, failover, and HA setups.
  • Monitor system health and maintain platform stability.
  • Ensure VoIP security: SIP fraud prevention, firewalls, fail2ban, ACL.
  • Document architecture, configurations, and system changes.
  • Participate in planning and scaling the voice platform.
📩 Requirements
Technical:
  • Strong knowledge of SIP, RTP/RTCP, and NAT traversal.
  • Hands-on experience with Asterisk (dialplan, pjsip/sip.conf, queues, AGI/AMI, call recording) or FreeSWITCH (configurations, dialplan, modules, Sofia-SIP).
  • Solid Linux skills (Debian/Ubuntu/CentOS): networking, system services, logging.
  • Ability to analyze SIP traffic (tcpdump, Wireshark, sngrep).
  • Experience integrating telephony with external services or CRM.
  • Understanding of QoS principles and voice route optimization.
  • Experience with logging/monitoring tools (Elasticsearch/Kibana — a plus).
  • English level B1 or higher.
Soft Skills:
  • Ability to work under high load and resolve incidents quickly.
  • System thinking, independence, and responsibility.
  • Clear communication and strong documentation skills.
🎁 Nice-to-Haves
  • Knowledge of PHP or Python.
  • Experience developing modules or scripts for Asterisk or FreeSWITCH.
  • Experience with VoIP fraud prevention.
  • Experience building contact center or multi-line call solutions.
  • Basic understanding of WebRTC.
🎁 What We Offer
  • Fully remote work format.
  • Strong technical team and access to deep expertise.
  • Opportunity to influence product architecture and develop a high-load voice platform.
  • Growth within an international eSIM product company.
Apply now — join Yesim!
Навыки
  • Asterisk
  • SIP-телефония
  • SIP
  • VoIP
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