L2 Duty Technical Support Engineer

Sippy Software, Inc. is a leading developer of carrier grade (class 4 /
5) softswitch technology for next generation Internet Telephony (VoIP) providers.

Sippy Softswitch delivers a complete network management solution for VoIP carriers on a single platform. This call-control software package enables service providers to build scalable and reliable VoIP networks. Based on the Session Initiation Protocol (SIP), it offers a full array of call-routing capabilities to maximize performance for both small and large packet voice networks.

Sippy currently handles more than 4 billion minutes of traffic, across thousands of client networks in more than 100 countries worldwide, assisting new ITSPs to launch on a weekly basis and contributing significant advancements to many open-source SIP projects.

Sippy Software develops and operates cutting-edge telecoms services & software. At Sippy Soft we aim for two things: building/operating great communications software and services and providing in-depth help to our customers to make sure they are successful.

• Supporting Sippy Software clients
• Taking part in system maintenance, migration and upgrade activities
• Taking part in testing and education of new system features

• Strong written and oral English (intermediate or higher);
• Intermediate knowledge of Linux / Unix administration
• Working knowledge of TCP/IP and network administration on Unix systems
• Base knowledge of sql queries;

To apply for this position, please send a PDF copy of your CV and include a cover note in the body of your email. Applicants should avoid sending their cover letters as an attached document.

Sippy Software, Inc. is a leading developer of carrier grade (class 4 /
5) softswitch technology for next generation Internet Telephony (VoIP) providers.

Sippy Softswitch delivers a complete network management solution for VoIP carriers on a single platform. This call-control software package enables service providers to build scalable and reliable VoIP networks. Based on the Session Initiation Protocol (SIP), it offers a full array of call-routing capabilities to maximize performance for both small and large packet voice networks.