VoIP Developer (FreeSwitch - Asterisk) (Pune)

VoIP Developer (FreeSwitch - Asterisk) (Pune)

04 Oct
|
Ishan Technologies
|
Pune

04 Oct

Ishan Technologies

Pune

Job Purpose

We are looking for an experienced VoIP Developer with 4+ years of hands-on experience in FreeSWITCH and VoIP technologies. The candidate will be responsible for developing, configuring, integrating, troubleshooting, and maintaining high-performance telephony and communication solutions.

Key Responsibilities

FreeSWITCH Development & Administration

- Develop, configure, and maintain FreeSWITCH-based VoIP solutions.
- Configure SIP profiles, gateways, extensions, dial plans, IVR, call routing, and call flows.
- Develop and customize FreeSWITCH XML dial plans, ESL, APIs, and modules.
- Integrate FreeSWITCH with dialers, CRM, contact centre, and third-party applications.
- Monitor and optimize FreeSWITCH performance and production environments.

SIP & VoIP Integration

- Configure and troubleshoot SIP trunks, SIP gateways, SIP signalling, and RTP media.
- Integrate FreeSWITCH with SBC, Asterisk, FusionPBX, VICIdial, CUCM, EPBX, and third-party PBX systems.
- Handle SIP authentication, codec configuration, NAT traversal, and interoperability issues.
- Work with SIP, RTP, SRTP, WebRTC, TCP/IP, and UDP.

Troubleshooting & Production Support

- Troubleshoot call failures, one-way/no audio, dropped calls, registration issues, and routing problems.
- Diagnose latency, jitter, packet loss, codec mismatch, SIP signalling, and RTP issues.
- Analyze FreeSWITCH logs and SIP traces for root-cause identification.




- Perform RCA and implement preventive solutions for recurring incidents.
- Coordinate with telecom carriers, network teams, and third-party vendors.

Qualification & Experience

- Bachelor's degree in Computer Science, IT, Electronics, Telecommunications, or equivalent.
- 4+ years of professional experience in VoIP/Telephony development.
- Strong hands-on FreeSWITCH experience is mandatory.
- Solid understanding of SIP, RTP, VoIP protocols, call flows, and telephony architecture.
- Experience with Asterisk, FusionPBX, VICIdial, SBC, or CUCM will be an added advantage.

Required Technical Skills

- FreeSWITCH – Mandatory
- SIP / SIP Trunking / SIP Profiles
- RTP / SRTP / WebRTC
- FreeSWITCH XML Dial Plan
- ESL (Event Socket Library) & FreeSWITCH APIs
- IVR & Call Flow Development
- SIP Gateway & PBX Integration
- SBC Configuration & Integration
- Linux / Linux CLI
- TCP/IP / UDP / NAT
- Codec Configuration & Troubleshooting
- SIP Trace & VoIP Debugging
- MySQL / PostgreSQL
- Python / PHP / Node.js or similar programming knowledge

Key Competencies

- Strong analytical and troubleshooting skills.
- Ability to design and implement complex VoIP call flows.
- Strong understanding of telephony and networking concepts.
- Experience handling production-critical VoIP environments.
- Good documentation and RCA skills.
- Ability to work independently and coordinate with internal teams, carriers, and vendors.

📌 VoIP Developer (FreeSwitch - Asterisk) (Pune)
🏢 Ishan Technologies
📍 Pune

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