Voip Asterisk Engineer & Developer

Voip Asterisk Engineer & Developer
Webarentor Ltd, Pakistan

Experience
3 Years
Salary
35,000 - 95,000 PKR
Job Type
Job Shift
Job Category
Traveling
No
Career Level
Telecommute
No
Qualification
Voip Telecoms & Asterisk
Total Vacancies
1 Job
Posted on
Mar 23, 2022
Last Date
Jun 23, 2022
Location(s)

Job Description

We looking for an enthusiastic VoIP Software Developer to join our growing R&D team, where you will be designing and developing new and existing voice services to be delivered via an award-winning global voice platform.

As a seasoned C/C++/Java/Go developer with knowledge/understanding of SIP and RTP, you will be applying your experience to create and maintain highly-reliable and scalable VoIP solutions using open-source SIP and PBX applications such as FreeSWITCH and OpenSIPS.

Applicants who can demonstrate their ability to extend server-based applications via third-party APIs, have experience with real-time media streaming and have built applications on cloud infrastructure will be highly suited to this role.

Responsibility:

• Develop and deploy new voice applications and the maintenance & diagnosing of any issues within existing applications.

• Translate requirements and designs into high-quality, secure code

• Implement and maintain automated unit and functional tests where appropriate

• Create utilities to aid testing & diagnosing issues

• Debug internally and externally-reported issues, and take both individual and collective responsibility to maintain optimal performance of applications at all times

• Work within the agile team and participate fully in all team meetings, sharing knowledge with the team and wider department

• Participate in peer code reviews, both as reviewer and reviewee

• Keep abreast of the latest security vulnerabilities, and develop with security in mind

• Demonstrate a willingness and motivation to learn and undertake self-initiated training

Job Specification

Asterisk PBX

Freeswitch

MYSQL

Linux

• C / C++

Java

Go Lang or similar

Knowledge of key voice protocols including SIP, RTP

Experience with extending FreeSWITCH modules, writing dial plans, and building Lua or Javascript scripts, or similar using Asterisk

• Configuration and customisation of open source SIP Proxies such as OpenSIPS, OpenSER, or Kamaillo

• Experience with migrating and deploying applications into AWS

• Knowledge of common media file formats such as WAV, MP3, PCM

• Experience with telecoms protocols such as ISDN and SS7

Job Rewards and Benefits

Webarentor Ltd

Computer & Network Security - Chelmsford, United Kingdom
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