Senior VOIP Developer

Adit; India; Presencial; INR 1.800.000 - 3.200.000; Jornada completa

Descripción de la vacante

Adit is seeking a Senior VOIP Developer with deep hands-on experience in Asterisk, FreeSWITCH, and SIP-based telephony. The role focuses on configuring, deploying, troubleshooting, and optimizing VoIP infrastructure, with emphasis on PBX systems, dialplan/XML programming, IVR design, and call center solutions.

You will work with Kamailio/OpenSIPS, VicidialNow, and OSDial, collaborating with development and operations to ensure security, uptime, backups, and comprehensive documentation.

Formación

• Hands-on experience with Asterisk/FreePBX, SIP-based telephony, and VoIP systems.

Responsabilidades

• Configure, deploy, maintain, and troubleshoot Asterisk/FreeSWITCH VoIP systems.

Conocimientos

VoIP engineering

SIP-based telephony

Dialplan/XML programming

AGI scripting

IVR design

Call center design

SIP routing

Linux system administration

WebRTC

System monitoring

Herramientas

Asterisk

FreeSWITCH

Kamailio

OpenSIPS

VicidialNow

OSDial

ASTPP

Lua

Perl

PHP

Node.js

Erlang

CURL

memcache

Linux

LAMP Stack

Descripción del empleo

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.

• 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.

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 Kamailio, OpenSIPS, Dialer/Open Source Support. Experience with VicidialNow, OSDial, ASTPP Billing Software, Development/Scripting. Familiarity with Lua, Perl, PHP, Node.js, Erlang, CURL, memcache, Infrastructure/Systems. Strong understanding of Linux System Administration, LAMP Stack, WebRTC, RabbitMQ, Database Administration.

Preferred: Database experience preferred: Oracle/OCP Certified is an added advantage.