As you can see in our profile that we are VoIP, Asterisk/Freeswitch, Kamilio/Opensip, WebRTC experts and have vast experience of over 14yrs in call, chat, audio, video, conference applications.
Technology, Platforms and languages used:
VoIP: Freeswitch, Asterisk, Twillio, Kamilio, FusionPBX, FreePBX, SIPJs
SIP Client: PJSIP, Csipsimple, doubango,
WebRTC: Native, Kurento, Tokbox, Jitsi, janus,
Web: Java, PHP, Laravel, Vue.js, Node.js, Angular, React.js, Django
Mobile: Swift, native, react native, Kotlin, Electron
Cloud Solution: AWS, Google cloud, Azure, Digital Ocean, Hetzner, Linux administration
Web Server: Nginx, Apache
DevOps: Github, Gitlab, Bitbucket, Jira, auth0, Jenkins, Kubernetes, Docker, Ansible
Below is the list and brief detail of some of our previous work. Hope you will look into our experience.
• Inbound/Outbound call center setup, development
• Sending voice mail as an attachment to user email.
• PBX Phone System with features like Call Forwarding, Call Transfer, Call Waiting, Call Park, Call Mute and many more.
• Calling card setup with all features like pinless dialing, speed dial, callerid etc.
• Made Customized IVRs for in various languages for different industries for survey, order taking etc.
• Auto Dialer/Predictive Dialer development
• FAX send/receive and integration
• Click to Call functionality
• Call/SMS Broadcasting with agents transfer option for call center solution.
• Hands on experience in codec management like G729, Opus, Ulaw/Alaw and many more
• Wrote many custom script in Perl/LUA for different call functions/scenario.
• CRM integration like vTiger, Mercury and others.
• Customized real time billing with features like active calls, real time call billing, provider management, rate card etc.
• Server load balancing, High Availability, 24*7 Support
• 3rd Party APIs integration like Twillio, Nexmo, Bandwidth and others for call, chat, video etc in applications.