{"id":203085,"url":"https://alion.io/job/curemd-senior-voip-engineer-voip-operations","title":"Senior VoIP Engineer — VoIP Operations","company":{"id":96135,"name":"CureMD","domain":"curemd.com","url":"https://alion.io/company/cure-md","size_band":"1001-5000","is_staffing_agency":false,"employer_type":"direct","is_intermediary":false,"listed_via":null,"ats_vendor":"Workday","truth_index":{"grade":"B","score":81,"open_postings":18,"ghost_share":0,"stale_share":0.778,"repost_share":0,"time_to_fill_p50_days":28,"computed_at":"2026-10-26T05:57:25Z"}},"role":"DevOps","role_family":"DevOps","seniority":"senior","employment_type":"full_time","work_mode":"on_site","remote_scope":null,"remote_scope_basis":null,"remote_working_hours":null,"hiring_geo_confidence":"structured","locations":["Lahore, Pakistan"],"countries":["PK"],"hiring_countries":[],"hiring_countries_total":0,"salary":null,"salary_estimate":{"min_usd":15000,"max_usd":42000,"period":"year","method":"global_role_cell_scaled_by_country","sample_n":3508},"experience_years_min":3,"visa_sponsorship":false,"relocation_package":false,"has_equity":false,"technologies":[{"name":"Amazon CloudWatch","optional":false},{"name":"Amazon EC2","optional":false},{"name":"Amazon S3","optional":false},{"name":"AWS","optional":false},{"name":"Bash","optional":false},{"name":"Grafana","optional":false},{"name":"Laravel","optional":false},{"name":"Linux","optional":false},{"name":"Nagios","optional":false},{"name":"Python","optional":false},{"name":"Rest API","optional":false},{"name":"Speech Recognition","optional":false},{"name":"WebRTC","optional":false},{"name":"Zabbix","optional":false},{"name":"PHP","optional":true}],"status":"live","first_seen_at":"2026-08-03T00:00:00Z","employer_posted_date":"2026-08-03","last_verified_at":"2026-10-25T02:12:58Z","board_verified":true,"closed_at":null,"days_open":84,"trust":{"level":"ok","repost_count":null,"flags":[],"days_open":84},"description":"Job Overview:\nThe Voice and Communications Engineer will operate, secure, and optimize enterprise VoIP, contact center, and messaging platforms across Asterisk/FreePBX, hosted PBX, carrier, and AWS environments. The role includes SIP and RTP troubleshooting, voice-quality monitoring, automation, API and CRM integrations, messaging compliance, and Tier-2/Tier-3 production support. The engineer will also support high availability, disaster recovery, security hardening, and continuous improvement of voice and messaging services.\nJob Responsibilities:\nOperate, maintain, and harden production Asterisk/FreePBX clusters and PBX environments, including dialplans, queues, IVRs, trunks, and recording.\nAdminister and support new deployments and the hosted GoToConnect PBX - provisioning, extensions, call flows, trunk configuration, and end-user troubleshooting.\nManage SIP trunks and PSTN connectivity with carriers - provisioning, routing, LCR, DID/TFN management, and incident coordination.\nMonitor call-quality KPIs (ASR, ACD, MOS, jitter, packet loss, PDD) and lead remediation when SLAs are at risk.\nOperate VoIP and supporting workloads on AWS - EC2, VPC, security groups, Route 53, S3, CloudWatch, and related services.\nMaintain network reachability, NAT traversal, firewall, and QoS configurations to ensure stable RTP/SIP traffic.\nManage backups, snapshots, patching, and disaster-recovery procedures for voice infrastructure.\nBuild and maintain integrations between Asterisk and Custom in-house VOIP Portals and Dynamics 365 - click-to-call, screen-pop, call logging, contact sync, and CDR push.\nIntegrate telephony with internal and third-party AI systems (voicebots, transcription, sentiment analysis, AI dialers, IVR automation).\nOperate and maintain messaging services across 10DLC SMS, longcode, shortcode, and fax, including carrier registration, brand/campaign compliance, throughput management, and DLR reconciliation.\nMaintain and support in-house web applications built on Laravel that interact with the telephony platform - bug fixes, minor enhancements, deployment, and operational support (deep Laravel development is not required, but the ability to navigate, debug, and modify the codebase is essential).\nWrite Python and Bash automation for provisioning, configuration management, log parsing, CDR processing, health checks, and recurring operational tasks.\nDevelop scripts and small tools that integrate Asterisk AMI/ARI, REST APIs (HubSpot, Dynamics, 3CX, GoToConnect, SMS gateways), and internal services.\nBuild dashboards, alerting, and runbooks using tools such as Grafana, Zabbix/Nagios, Homer, sngrep, and CloudWatch.\nReduce manual operations through automation - provisioning, user onboarding/offboarding, trunk failover, and routine audits.\nHarden VoIP and messaging infrastructure against toll fraud, SIP scanning, brute-force attempts, and abuse; manage TLS/SRTP, fail2ban, ACLs, and rate limits.\nEnsure adherence to 10DLC, CTIA, and TCPA requirements for messaging traffic, and applicable regulations for voice (E911 where relevant).\nMaintain accurate, version-controlled technical documentation, network diagrams, runbooks, and SOPs\nAct as Tier-2/Tier-3 escalation for voice, messaging, and integration incidents.\nPartner with development, NOC, and contact-center teams on rollouts, migrations, and continuous improvement.\nParticipate in an on-call rotation for production voice and messaging services.\nQualifications:\nBachelor’s or Master’s degree in Computer Science, Software Engineering, or a related field.\n3-5 years of experience in enterprise voice engineering, working with Linux-based telephony platforms and scripting or programming languages for automation and system integration.\nGood understanding of Linux environments, including shell scripting, configuration, and system backups.\nStrong experience with Asterisk, FreeSWITCH, Kamailio/OpenSIPS, SIP proxies, and other open-source VoIP platforms.\nWorking knowledge of WebRTC-based voice & video communication and open-source VoIP ecosystems.\nDeep understanding of SIP signaling, RTP/SRTP, DTMF, codecs, NAT traversal, and fax (T.38/Passthrough) troubleshooting.\nStrong knowledge of SIP trunking, carrier interconnects, DID provisioning, number portability, E911, and telecom regulations.\nExperience implementing high availability (HA), redundancy, failover, disaster recovery, and business continuity solutions for mission-critical voice platforms.\nAbility to interpret system diagrams, architecture blueprints, call flows, and carrier workflows across on-premises and cloud environments.\nExcellent verbal and written communication skills required for cross-functional coordination and vendor interaction.\nCompensation and Benefits:\nFinancial:\nCompetitive salary and bi-annual bonus.\nFast track and uncapped career growth for high performers.\nCompany-sponsored vehicle financing (car and bike).\nInterest-free loans.\nProvident Fund: CureMD matches up to 8% of your base salary.\nHealth and Wellness:\nIn-house clinic with a team of certified male and female doctors with 24/7 telemedicine service.\nHospital treatment monitoring by company doctors.\nComprehensive health coverage for your immediate family (outpatient, inpatient, maternity and parents' inpatient).\nExclusive health benefits and discounts at top class clinics and labs.\nSupportive Workplace:\nPick-up and drop-off services for female employees.\nIn-house daycare facility.\nIn-house gym and recreational area to unwind.\nContinued Learning:\nCompany-sponsored trainings, workshops, development programs and retreats.\nPaid specialized trainings/certifications.\nThe Difference You’ll Make:\nAt CureMD, every role, whether senior or junior, plays a pivotal part in transforming healthcare. By joining our innovative team, you’ll contribute to groundbreaking technology that directly impacts patient care, enhances healthcare efficiency, and saves lives globally. Your skills and passion will drive meaningful change, helping us deliver solutions that support healthcare professionals in critical, real-time settings. Together, we’re not just advancing technology - we’re making a tangible difference in people’s lives. 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