Job Description

Responsibilities:
Design and develop scalable solutions to deliver reliable SIP services to a multitude of distributed endpoints including on premise Call Processing Servers, SIP proxy servers, handsets, and soft-phones.
Enhance and maintain existing code and participate in design reviews and code reviews with other team members.
Create custom provisioning tools to automate configuration of voice stack
Participate in projects intended to improve infrastructure, including the integration of current systems and software
Draft technical documentation
Facilitate knowledge transfer to deployment and support teams
Work with the Production Services and Support leaders towards keeping Service Availability above target.
Other duties as assigned by management

Tools and Technologies used include:
SIP protocol and its use in VOIP Services
SIP proxies and/or SIP PBXs preferably open source products like Kamailio (OpenSER), FreeSWITCH, Asterisk, and FreePBX etc.
Networking with Switches, Routers, Firewalls, IPSs and Security Appliances




Programming experience with PHP / Perl / JAVA / C and custom modules with open source products
Relational Databases ( MySQL), JDBC / ODBC
Linux server operations, deployment and performance monitoring and shell scripting Qualifications

Qualifications:

Education / Training:

University degree in Computer Engineering, Computer Science or equivalent.

Mandatory Skills & Experience:
Experience coding and debugging Asterisk source code.
Highly proficient in PHP, Perl and MySQL
Knowledge of software design best practices.
Proficient using source code management systems.
Proficient understanding of SIP, MGCP, H323.
Solid understanding of a hosted, highly available setting.
Proficient with configuring Asterisk, including writing dialplan and custom application code.
Knowledge of common configuration methods of endpoints (TR-069, HTTPS, TFTP).
Working knowledge of T1/E1/J1 interfaces, PR

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