SIP/Dialer Software Engineer (Asterisk/Freeswitch)

Job Title: SIP/Dialer Software Engineer (Asterisk/Freeswitch)

Location: Gurgaon

Experience Level: 5+ Years

Job Summary

We are seeking an experienced SIP/Dialer Software Engineer with expertise in configuring

and maintaining Asterisk or Freeswitch systems. The role involves working with SIP servers,

call center dialers, and VoIP technologies to ensure seamless communication, system reliability,

and optimal performance.

The ideal candidate will be instrumental in configuring, optimizing, and troubleshooting dialer

systems while collaborating with cross-functional teams to support call center operations. If you

are passionate about telephony systems and possess in-depth knowledge of SIP, VoIP, and

dialer technologies, we encourage you to apply.

Key Responsibilities

• SIP Server Configuration & Management: Install, configure, and manage SIP servers

(e.g., Axterisk, Freeswitch, Kamailio etc) to ensure seamless integration with call center

systems and stable SIP trunking.

• Dialer Setup & Optimization: Configure, maintain, and optimize outbound dialer

systems (predictive, progressive, manual, or preview modes) to meet business goals

and compliance standards.

• Asterisk/Freeswitch Expertise: Provide expert-level support, installation, and

troubleshooting for Asterisk or Freeswitch systems, ensuring high system availability.

• System Monitoring & Maintenance: Continuously monitor system performance,

conduct root cause analysis, and proactively resolve issues to prevent disruptions.

• VoIP Performance Enhancement: Analyze call quality metrics, reduce latency, and

optimize RTP/SIP signaling flows to improve user experience.

• Integration with Business Systems: Develop and maintain integrations between the

dialer system, CRMs, databases, and third-party telephony platforms.

• Collaboration & Support: Collaborate with teams including network engineers and

developers to resolve issues and provide technical support to users and agents.

• Security Compliance & Updates: Ensure systems are updated with the latest patches,

security enhancements, and regulatory compliance measures.

• Documentation & Knowledge Sharing: Maintain detailed records of configurations,

updates, and troubleshooting steps, and share knowledge across teams.

Required Skills & Qualifications

• SIP Server Expertise: Proficient in setting up and managing SIP servers such as

Asterisk, Freeswitch, Kamailio, or similar platforms.

• Hands-On Asterisk/Freeswitch Experience: At least 2+ years of working experience

with configuration, maintenance, and optimization.

• Dialer Systems: Proven experience with call center dialers, including configuration of

predictive/manual/progressive dialing modes.

• VoIP Protocols: Strong understanding of SIP, RTP, WebRTC, and other VoIP

technologies.

• Network Knowledge: Familiarity with NAT traversal, QoS, and basic network

troubleshooting for VoIP systems.

• Programming Skills: Working knowledge of scripting or programming languages such

as Python, Java, or Bash for automation and system configuration.

• Database Integration: Experience with SQL/NoSQL databases for storing and

managing call logs and telephony metadata.

• Analytical & Problem-Solving Skills: Ability to analyze logs, identify bottlenecks, and

implement solutions efficiently.

• Excellent Communication: Strong written and verbal communication skills to convey

technical information effectively.

Preferred Qualifications

• Experience with cloud-based telephony solutions
• Familiarity with high-availability systems and clustering techniques for telephony

platforms.

• Knowledge of compliance standards in call center operations, such as TCPA, TRAI

regulations, or similar.

• Experience with load testing and scalability for telephony systems.

Education

• Bachelors degree in Computer Science, Telecommunications, or a related field, or

equivalent work experience.