Sr. VOIP Developer - Remote (US Shift)

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

Job summary

Adit is seeking a Sr. VOIP Developer with deep hands-on experience in Asterisk, FreeSWITCH, SIP-based telephony, and open-source dialers. The role covers configuration, deployment, troubleshooting, and optimization of VoIP infrastructure and communication platforms.

The candidate should excel in PBX systems, dialplan/XML programming, IVR design, and call center solutions, with exposure to Kamailio/OpenSIPS and SIP routing. Strong Linux server management and security practices are required.

Qualifications

• Senior VoIP developer with deep hands-on VoIP and dialer experience.
• Strong expertise in Asterisk/FreeSWITCH, SIP-based systems, XML and AGI scripting.
• Experience with IVR, call centers, and open-source dialers.

Responsibilities

• Configure, deploy, and troubleshoot Asterisk/FreeSWITCH based VoIP systems.
• Migrate data, perform upgrades, and optimize system performance.
• Develop dialplans, call flows, and telephony logic.
• Design IVR/IVRS, auto-reminders, and voice broadcasting setups.
• Support call centers, routing logic, and conferencing environments.
• Manage SIP trunking and ensure voice quality and failover.
• Work with Kamailio/OpenSIPS-based routing and WebRTC integrations.
• Collaborate with developers and operations for deployments.

Skills

Asterisk

FreeSWITCH

SIP/VoIP

Dialplan Programming

XML Programming

AGI Scripting

IVR Design

Call Center Setup

VicidialNow

OSDial

ASTPP Billing

SIP Proxy / Routing

Job description

We are looking for a highly skilled Sr. 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 |

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

Working knowledge of:

Experience with:

VicidialNow | OSDial | ASTPP Billing Software | Development / Scripting |

Familiarity with:

Strong understanding of:

Database experience preferred:

Oracle / OCP Certified is an added advantage