Senior VoIP Telephony Engineer (FreeSWITCH)

Background: Afghanistan Holding Group (AHG) provides services to government, international organizations, and private sector clients across Afghanistan, including research, monitoring, and evaluation services. AHG has and will establish E1 connections with the mobile network operators to route voice traffic via SIP connections to our cloud-based application servers, and run a local IVR, in addition to local SIP phones. We are currently seeking an IP Telephony Engineer with extensive skills and experience to deploy, operate and maintain a Plivo PBX server (FreeSWITCH-based) in our Kabul office. Duties & Responsibilities: • Implement and maintain VoIP telephony server using a locally installed Plivo (FreeSWITCH-based) server. • Connect existing E1 connections (via SIP over VLAN from operator) with local Plivo server. • Connect cloud-based IVR application server (Twilio, Plivo or similar) with local Plivo server. • Support access to the Plivo server using it’s RESTAPIs or RESTXML from cloud-based RapidPro application server. • Connect local SIP phones to local Plivo server. • Recommend and configure low-cost call center management software for small call center using local SIP phones. • Establish and test new carrier connections. Work with carriers to diagnose and fix issues. • Develop process and procedures for implementation and operational support of systems. • Verify, test and document all call routes, configurations, and provide guidance on backups and system security. Skills & Specifications: • Strong engineering and operational experience in service provider networks. • In-depth understanding of SIP, RTP and RTCP • Experience configuring and maintaining open source SIP server software including patching, upgrades, logs, debugs and virtual traces. • In-depth understanding of open source PBX features, configuration, deployment and support (Asterisk or FreeSWITCH). • Knowledge of network server policies, routing network elements and collocation modeling. • Ability to perform SIP debugs, capture and analyze SIP and RTP to isolate and resolve issues • Experience working directly with carriers and trunk providers. • In-depth understanding of switching for a telephony standpoint into the PSTN Qualifications: • BS in Computer Science, Electrical Engineering, Telecommunications Engineering or equivalent required • Minimum 7+ years experience related to VoIP telephony engineering • Experience with FreeSWITCH or Asterisk is required • Expert knowledge on Linux