Senior VOIP Developer

Adit; India; On-site; INR 1,800,000 - 3,200,000; Full time

Job summary

Adit is seeking a Senior VOIP Developer to lead design, deployment, and optimization of VoIP infrastructure. You will configure Asterisk and FreeSWITCH, craft dialplans, and implement IVR solutions. Strong focus on SIP routing, WebRTC, and scalable telephony platforms.

Ideal candidates have deep experience with Kamailio/OpenSIPS and open-source dialer ecosystems, plus governance of QA, backups, and security in Linux-based environments.

Qualifications

• Deep hands-on experience with Asterisk and FreeSWITCH platforms.
• Proficient in SIP, VoIP, and telephony protocols.
• Strong dialplan/XML programming and AGI scripting.
• Experience building IVR solutions and call center deployments.

Responsibilities

• Configure and maintain Asterisk/FreeSWITCH VoIP systems.
• Migrate data and upgrade VoIP platforms; optimize performance.
• Develop dialplans and XML-based call routing logic.
• Design IVR/IVRS solutions and automated voice reminders.

Skills

Asterisk

FreeSWITCH

SIP

Dialplan

XML

AGI Scripting

IVR Design

Call Center

WebRTC

Kamailio

OpenSIPS

Tools

Linux Servers

Job description

We are looking for a highly skilled Senior VOIP Developer with strong hands-on experience in Asterisk, FreeSWITCH, SIP-based telephony systems, and open-source dialer ecosystems. The ideal candidate will be responsible for the configuration, customization, deployment, troubleshooting, and optimization of VoIP infrastructure and communication platforms.

This role requires deep technical expertise in PBX systems, dialplan/XML programming, IVR design, call center solutions, and SIP server technologies such as Kamailio and OpenSIPS.

Key Responsibilities

• Configure, deploy, maintain, and troubleshoot Asterisk / FreeSWITCH based VoIP systems
• Perform Asterisk / FreePBX data migration, upgrades, and system optimization
• Develop and manage Dialplans, XML-based call flows, and telephony logic
• Design and implement IVR / IVRS solutions, auto-reminder systems, and voice broadcasting setups
• Build and support call center environments, conference bridges, inbound/outbound calling systems, and routing logic
• Work on AGI scripting and telephony automation to improve system efficiency
• Configure and manage SIP trunking, call routing, failover, and voice quality optimization
• Deploy, maintain, and troubleshoot open-source dialer platforms such as VicidialNow and OSDial
• Support and optimize Kamailio / OpenSIPS based SIP routing and proxy environments
• Work with WebRTC, voice logger systems, and real-time communication solutions
• Monitor VoIP infrastructure performance and proactively resolve service issues
• Collaborate with development and operations teams for integrations, deployments, and enhancements
• Manage Linux-based servers and support backend telephony applications
• Maintain system security, uptime, backups, and documentation

Required Skills & Technical Expertise

Core VoIP / Telephony

Strong hands-on experience with

• Asterisk
• FreeSWITCH
• SIP / VoIP
• Dialplan Programming
• XML Programming
• AGI Scripting
• IVRS / IVR Design
• Call Center Setup

Additional skills

• Conference Systems
• Voice Broadcasting
• Auto Reminder Systems
• Voice Logger
• SIP Proxy / Routing

Oracle / OCP Certified is an added advantage