13+ SIP Jobs in India

What will you do?

• Oversees testing, deployment & maintenance of the Voice infrastructure.
• Performing a variety of tasks associated with user provisioning and call routing within the assigned platform 
• Build loaders for internal systems ensuring tasks are completed accurately and in a timely manner
• Participate in meetings with clients and internal departments to understand the requirements of call routing
• Document and design call flow diagrams to ensure that all call variables have been accounted for in preparing a routing plan
• Support the clients solution by correcting any issues with call routing or provisioning during the initial implementation phase, constantly communicating status of any issues with client
• Complete testing to ensure that call flow is operating properly
• Monitors and ensures compliance to standards, policies, and procedures

What are we looking for?

• 0.6~2 year’s implementation experience in Asterisk PBX, IP Telephony, SIP.
• Able to thrive in a demanding and team-oriented environment
• Basic and strong knowledge of the Linux operating system.
• Knowledge of bash and perl shell scripting is an added advantage.
• RHCE Certification is an added advantage.

Job Title:

Telephony Engineer

Job Description:

- Support the operations team with debugging issues related to the telephony platform

- Identify issues with calls using monitoring and analysis tools such as VoIPMOnitor

- Write scripts to automate tasks, monitor the services and functionalities

- Contribute to the improvement of the system by providing ideas

- Help with capacity planning

- Perform the scheduled maintenance activities

- Modify the existing code to accommodate new features

Experience Range:

3- 6 years

Educational Qualifications:

Any graduation,

Job Responsibilities:

  • Knowledge of open source technologies, VoIP, SIP, WebRTC etc.

  • Shell scripting.

  • Experience in Asterisk and Kamailio or OpenSIPS.

  • C/C++ or JS programming with Linux.

  • 5-8 years experience.

  • Good communication skills.

  • Willingness to work in the night, whenever required, for handling support related issues.

Skills Required:

ViOP, OpenSIPS, SIP, Shell Scripting, Asterisk, kamailio, C++, Linux, Development,

About Us 

Rezo.ai is an AI-Powered Contact Centre that enables enterprises to enhance customer experience and boost revenue by automating and analyzing customer agent interactions across multiple channels including voice, email, chat/WhatsApp, and social, at the required scale, whilst training agents with minimal costs 

How do we do it 

Rezo’s AI-Powered contact center leverages ground-breaking technologies in AI, ML, ASR, NLP, RPA, and predictive intelligence to transform customer experience and reduce costs by automating, and analyzing social media, whilst coaching them. 

Job Description

• Preference would be given to having experience in IVR and Call Center application development.
• Hands-on experience in Kamailio SIP Server
• Hands-on experience in WEBRTC
• Asterisk integration with PSTN is preferred.
• Asterisk integration with Database (MySQL and MSSQL) with knowledge of queries and stored procedures.
• Scripting with Shell, Perl.
• Experience in installing Hosted VoIP
• 
  Nice to have
• Good knowledge of SIP and IAX.
• MySQL and Telephony Domain.
• Ability to test and troubleshoot Asterisk applications/issues
• Knowledge of C programming in Asterisk

Qualifications

Candidate Profile
The candidate must have:

• 2 - 4 years of experience working on the Asterisk platform.
• Strong Knowledge of PBX, SIP, and RTP protocols.
• Strong knowledge of multi-asterisk server architecture
• Sound knowledge of Asterisk Dial-plan, Asterisk Gateway Interface and Asterisk Manager Interface.
• Worked on Queue, IVR and Voicemail-related applications.
• Expert in Asterisk installation, configuration and troubleshooting.
• Expert knowledge of installing and configuring Asterisk with PSTN card.
• R & D skills in Asterisk.
• Experienced with Linux (LAMP) servers (Centos, Apache, MySQL, PHP)

Desired Candidate Profile

• Minimum 3 years of hands-on development experience
• Experience with VoIP based technologies based on SIP protocol.
• Solid C, C++ experience, Python
• Experience in developing multi-threaded and client server application
• Experience in Linux/Unix environment
• Experience/Knowledge in at least three of the following technical areas is also required:
• Object oriented design and methodologies
• Experience developing cloud-based services (e.g. SaaS, PaaS)
• Knowledge and development background in web services, or web based technologies using REST.
• Experience with highly available/redundant softwer.
• Strong communication skills required.
• Strong writing skills required.
• Ability to multi-task and handle changing priorities.
• Knowledge of software development methodology, build tools and product life cycle; experience in agile development is a plus.
• Excellent team skill, can-do attitude, focus on quality.

Required Skills: GSM, Linux, Networking, infrastructure, VOIP, PHP, Perl, SIP, troubleshooting, Python,HTML

• 5+ years of solid experience as an Asterisk Developer
• Must be proficient at developing software using a scripting language. A distinct advantage if you have experience using Python
• Sound knowledge of Asterisk Installation, Configuration, Dialplan, Call troubleshooting (SIP, ISDN)
• Solid experience of working With Linux Operating Systems
• Good understanding of VoIP, SIP, SS7.

Monitor, analyze and fix issues for consumer and business VoIP services and endpoints.

Roles and Responsibilities:

• As a senior engineer, take complete ownership of supporting customer VoIP deployments by debugging issues and providing fixes.
• Document existing call flows
• Build customer confidence to grow/lead local team
• Support customer during PST timezone during critical releases

Required Skills:

• 5+ yrs of supporting VoIP services/applications on Linux platform
• Strong scripting skills with bash/python/perl
• Expertise in SIP call flow analysis and debugging using Wireshark
• Experience in debugging Kamailio and Freeswitch/asterisk based applications is a must
• Good problem solving/analytical skills
• Excellent written and verbal communication

Preferred Skills:

• Experience working with open source projects
• Exposure to Level 3-Carrier Integration
• Knowledge of networking protocols