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 crossfunctional teams to support call center operations. If you
are passionate about telephony systems and possess indepth 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 expertlevel 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 thirdparty 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.
HandsOn 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 & ProblemSolving 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 cloudbased telephony solutions
Familiarity with highavailability 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
Bachelor s degree in Computer Science Telecommunications or a related field or
equivalent work experience.
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