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الوصف الوظيفي Job Responsibilities: · Understand how all Telephony services work and how they are integrated to the platform. · Operate, maintain, expand and support SIP proxies based on Kamailio or OpenSIPS applications. · Develop new functionality with currently deployed solutions. · Handle level 3 troubleshooting escalations and triaging. · Analyze telephony traffic patterns and identify issues and anomalies. · React to critical alerts in order to rapidly return to a full-service state. · Troubleshoot and resolve voice and network protocol communication issues. · Design, build, test, deploy and maintain monitoring, alerting, QA and logging tools for Telephony applications. Coordinate system maintenance and deployment events. المهارات · Minimum of 3 years’ experience (networking and VoIP) required, 5 years preferred · Excellent verbal and written communication skills · Working knowledge of TCP/IP, SIP · Knowledge of Linux (CentOS, Debian): Security, Bash, TCP/IP, High Availability, Package management (yum, apt) · Experience with Asterisk with AGI and customized PBX. · Experience with Freeswitch and AGI and customized PBX. · Experience with VOIP gateways/SBC like OpenSips and/or Kamailio. · Hand on experience with Mysql Database. · Programming skills in bash, perl, php, python · Understanding of NAT, PAT, Firewall issues as they relate to VoIP · Experience in interfacing with call routing / rating / billing systems · Strong interpersonal skills, positive attitude, work well in teams · Ability to write scripts in Perl, PHP, Python and bash تفاصيل الوظيفة https://www.bayt.com/ar/uae/jobs/voip-developer-4500734/ |
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