I am a Senior VoIP Engineer and UC Architect with extensive hands-on development experience. I have built and deployed a web management platform for Asterisk multi-tenant IP-PBX (Centrex) and a web UI for Kamailio to manage routing, trunks, numbers, and number-to-backend mapping. Currently, I operate and support approximately 40 standalone PBX servers and 6 multi-tenant Centrex platforms, including API integrations with customer infrastructures.
My expertise includes working with Asterisk components such as PJSIP/chan_sip, dialplan, AMI/AGI, ODBC RealTime, queues, CDR/CEL, and real-time dashboards using Redis and SSE/WebSocket technologies. I also design and support SIP proxy logic in Kamailio, implementing multi-tenant routing patterns, dispatcher-based load balancing, and clear inbound/outbound routing separation.
I develop real-time monitoring components and integrate them into PHP-based admin UIs. Additionally, I create reporting and analytics solutions using MariaDB schemas, triggers, views, and queries for queue_log and CDR data, including data retention and archiving strategies.
I implement AI voice modules including Google Speech-to-Text, Text-to-Speech, and LLM-based intent and answer classification with multilingual call flows supporting Russian, Lithuanian, Latvian, and Estonian. I store transcripts and results in databases for audit and automation purposes.
I communicate fluently in Russian and Lithuanian, with basic written English and limited speaking ability, preferring written communication. I am comfortable working remotely or in hybrid environments within the Baltics region.
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Built and deployed a web management platform for Asterisk multi-tenant IP-PBX (Centrex), enabling tenant administration including companies, accounts, numbers, routes, queues, and reporting. Developed a web UI for Kamailio to manage routing, trunks, numbers, and number-to-backend Asterisk mapping for centralized SIP routing. Operate and support approximately 40 standalone PBX servers and 6 multi-tenant Centrex platforms in production, including customer API integrations and multi-server interconnections. Implement and maintain Asterisk call flows including contexts, dialplan logic, SIP trunks, queues, call recording, CDR/CEL, and incident troubleshooting for SIP/RTP. Design and support SIP proxy logic in Kamailio with multi-tenant routing patterns, dispatcher-based balancing, and inbound/outbound routing separation. Develop real-time monitoring components using AMI listeners, Redis, and SSE/WebSocket integrated into PHP-based admin UI. Create reporting and analytics with MariaDB schemas, triggers, views, queries for queue_log/CDR, and retention/archiving strategies. Implement AI voice modules including STT/TTS pipelines, intent/answer classification, multilingual prompt logic, and database storage for transcripts and results.
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