Senior VoIP Developer
你将负责设计，构建和维护强大的可扩展的 VoIP 系统，包括软交换机，拨号器，PBX GUIs,账单解决方案以及实时集成。你必须在 Asterisk,FreeSWITCH,Kamailio,OpenSIPS,VICIdial,FreePBX,FusionPBX 等账单系统中拥有专业知识.GUI 开发，API 集成和多租户 VoIP 系统是必不可少的。要求关键责任：·使用 ○ Asterisk, FreeSWITCH, Kamailio, OpenSIPS ○ GUI 框架：FreePBX, FusionPBX ○拨号器：VICIdial, GoAutoDial ○ 发票平台：MagnusBilling, ASTPP ·设计和开发定制 IVR,调用路由逻辑和 SIP 干部集成。·开发基于网络的 VoIP 控制面板，仪表板和定制管理员/客户 GUI. ·与第三方平台集成如：○ WhatsApp Business API ○ CRM 和 SMS 网关系统 ○ 付款和电子邮件系统 ·执行 SIP 调试，音频故障排除，NAT 传输处理和代码编程优化。· 工作多项设置，可用性，失败架构和高安全性设置。· 自动化使用 Bash,Python 或 PHP 的脚本部署。·在 Proxmox,VMware 或云提供商如 AWS/DigitalOcean 上管理 VoIP 实例。保密要求技能和经验：·至少3年的 VoIP 系统开发经验。·强大的 SIP/RTP/NAT,VoIP 代码和信号知识。·与：○Linux 环境 (Ubuntu,Debian,CentOS) 亲身经验。○MySQL/PostgreSQL 数据库 ○Bash-Thone/PHP 编写。·工具：Wireshark,sngrep,tcpdump,Aster CLI,FS-CLI 优先技能 (Bonus):熟悉 WebRTC,STIRSHAK,并遵守 API. ·多方应用程序和协议集成平台的经验。· 之前在电信 SaaS,初创或 CPaaS 提供商工作。我们提供的好处：·基于技能和经验的竞争性工资。·远程工作灵活性和基于绩效的增长。·接触先进的电信项目和平台。·快速发展，协作和创新驱动文化。最初发布于喜马拉雅山。
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This is a remote position.
Location: Remote / Jaipur (Optional Onsite) Address: Sitapura, Jaipur Experience Required: 3+ Years in VoIP Development We are looking for an experienced and passionate VoIP Developer to join our growing tech team. You will be responsible for designing, building, and maintaining robust and scalable VoIP systems including softswitches, dialers, PBX GUIs, billing solutions, and real-time integrations. You must have expertise in platforms like Asterisk, FreeSWITCH, Kamailio, OpenSIPS, VICIdial, FreePBX, FusionPBX, and billing systems such as MagnusBilling and ASTPP. Experience with GUI development, API integrations, and multi-tenant VoIP systems is essential.
Requirements
Key Responsibilities:· Build and manage VoIP platforms using:
○ Asterisk, FreeSWITCH, Kamailio, OpenSIPS ○ GUI frameworks: FreePBX, FusionPBX ○ Dialers: VICIdial, GoAutoDial ○ Billing platforms: MagnusBilling, ASTPP
· Design and develop custom IVRs, call routing logic, and SIP trunk integrations.
- Develop web-based VoIP control panels, dashboards, and custom admin/client GUIs.
- Integrate with 3rd-party platforms like:
○ WhatsApp Business API ○ CRMs and SMS Gateways ○ Payment and Email Systems
· Perform SIP debugging, audio troubleshooting, NAT traversal handling, and codec optimization.
- Work on multi-tenant architecture, high availability, failover, and security setup.
- Automate and script deployments using Bash, Python, or PHP.
- Manage VoIP instances on Proxmox, VMware, or cloud providers like AWS/DigitalOcean. Confidential Required
Skills & Experience:
· Minimum 3 years of experience in VoIP system development.
- Strong knowledge of SIP/RTP/NAT, VoIP codecs, and signaling.
- Hands-on experience with:
○ Linux environments (Ubuntu, Debian, CentOS) ○ MySQL/PostgreSQL databases ○ Scripting in Bash/Python/PHP
· Tools: Wireshark, sngrep, tcpdump, Asterisk CLI, FS-CLI Preferred Skills (Bonus):
- Familiarity with WebRTC, STIR/SHAKEN, and telecom compliance.
- Experience in multi-tenant VoIP platform scaling.
- API development and third-party app integrations.
- Prior work in a telecom SaaS, startup, or CPaaS provider.
Benefits
What We Offer:
· Competitive salary based on skills and experience.
- Remote work flexibility and performance-based growth.
- Exposure to advanced telecom projects and platforms.
- Fast-paced, collaborative, and innovation-driven culture.
Originally posted on Himalayas