Senior VoIP Telephony Engineer (FreeSWITCH)

About Afghanistan Holding Group

Job Summary

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.

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.

Job Requirements

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

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

Submission Guideline

Interested and qualified candidates should send their CV to submission email address or applying online via Jobs.af
application form. Only short-listed candidates will be contacted for interview.
Please indicate the name of position in the subject line of your e-mail.

Functional Area

Engineering

Countries

Submission Email

careers@ahg.af

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