Job Summary
We are seeking an experienced specialist who can design develop and maintain Asterisk/FreePBX telephony systems for contact center and conferencing environments while also building APIs for call routing call bridging conferencing control and integrations.
This role combines VoIP engineering contact center architecture conferencing and backend/API development.
Key Responsibilities / Duties
1. Asterisk / FreePBX Engineering (Contact Center and amp; Conferencing)
Architect deploy and manage Asterisk/FreePBX for high-volume contact center and conferencing use cases.
Configure:
IVRs ACD skill-based routing multi-level queue structures
Outbound dialer workflows (predictive preview progressive)
Call conferencing (multi-party conferences PIN management moderator controls dynamic conference rooms)
Whisper barge monitor recording quality monitoring
Optimize call flows for low latency and high concurrency.
2. API Development and amp; Call Automation
Design build and maintain REST APIs for:
Call bridging (agentcustomer multi-party warm transfers blind transfers)
Conference creation joining and controls (mute/unmute kick lock/unlock recording control)
Click-to-call and amp; CRM-triggered calls
Dynamic call routing and amp; IVR adjustments
Agent login/logout pause/unpause
Call event streaming (webhooks)
Conferencing analytics (participant count duration event logs)
Work with Asterisk interfaces such as:
ARI (Asterisk REST Interface)
AMI (Asterisk Manager Interface)
AGI (Asterisk Gateway Interface)
3. Integrations
Integrate telephony systems with:
CRM platforms (Salesforce Zoho Freshdesk Dynamics HubSpot)
Conferencing portals or custom meeting management apps
Ticketing systems chat systems workforce management tools
Reporting and analytics dashboards
Sync conference metadata call logs queue stats and recordings.
4. Monitoring Performance and amp; Troubleshooting
Ensure high availability low jitter low packet loss and optimal VoIP performance.
Troubleshoot SIP and RTP issues using:
sngrep
tcpdump
Wireshark
Asterisk CLI logs
Optimize codecs transcoding performance and server load.
Monitor real-time:
Queue performance
Conference performance
SLA metrics
MOS score and call quality indicators
5. Security and amp; Reliability
Implement security measures for voice and conferencing systems:
Fail2ban IP filtering SIP firewalling
Anti-fraud and intrusion prevention
Manage system upgrades patches backups and DR plans.
Deploy HA clusters for mission-critical contact center conferencing environments.
Required Skills and amp; Qualifications
35 years hands-on experience with Asterisk and amp; FreePBX (contact center conferencing).
Strong understanding of SIP RTP conferencing modules bridge applications.
Proven experience building APIs and backend services ( Python PHP Go etc.).
Hands-on experience with ARI AMI AGI and dialplan logic.
Strong Linux skills (CentOS Debian Ubuntu).
Experience configuring:
Contact center queues and routing
Conference bridges
Recordings monitoring tools
Experience with SIP trunking and carrier integrations.
Preferred / Good-to-Have Skills
Experience with large-scale conferencing platforms (Asterisk ConfBridge MeetMe or custom conferencing solutions).
Knowledge of OpenSIPS/Kamailio for scaling conferencing or call center environments.
Familiarity with WebRTC and browser-based conferencing.
Knowledge of predictive dialers like VICIdial/GoAutoDial.
Cloud deployment experience (AWS/GCP/Azure).
Familiarity with microservices and distributed architectures.
Soft Skills
Strong analytical and debugging mindset.
Ability to document systems flows and API specs clearly.
Good communication and collaboration skills.
Ability to work in fast-paced contact center or conferencing environments.
Job SummaryWe are seeking an experienced specialist who can design develop and maintain Asterisk/FreePBX telephony systems for contact center and conferencing environments while also building APIs for call routing call bridging conferencing control and integrations.This role combines VoIP engineerin...
Job Summary
We are seeking an experienced specialist who can design develop and maintain Asterisk/FreePBX telephony systems for contact center and conferencing environments while also building APIs for call routing call bridging conferencing control and integrations.
This role combines VoIP engineering contact center architecture conferencing and backend/API development.
Key Responsibilities / Duties
1. Asterisk / FreePBX Engineering (Contact Center and amp; Conferencing)
Architect deploy and manage Asterisk/FreePBX for high-volume contact center and conferencing use cases.
Configure:
IVRs ACD skill-based routing multi-level queue structures
Outbound dialer workflows (predictive preview progressive)
Call conferencing (multi-party conferences PIN management moderator controls dynamic conference rooms)
Whisper barge monitor recording quality monitoring
Optimize call flows for low latency and high concurrency.
2. API Development and amp; Call Automation
Design build and maintain REST APIs for:
Call bridging (agentcustomer multi-party warm transfers blind transfers)
Conference creation joining and controls (mute/unmute kick lock/unlock recording control)
Click-to-call and amp; CRM-triggered calls
Dynamic call routing and amp; IVR adjustments
Agent login/logout pause/unpause
Call event streaming (webhooks)
Conferencing analytics (participant count duration event logs)
Work with Asterisk interfaces such as:
ARI (Asterisk REST Interface)
AMI (Asterisk Manager Interface)
AGI (Asterisk Gateway Interface)
3. Integrations
Integrate telephony systems with:
CRM platforms (Salesforce Zoho Freshdesk Dynamics HubSpot)
Conferencing portals or custom meeting management apps
Ticketing systems chat systems workforce management tools
Reporting and analytics dashboards
Sync conference metadata call logs queue stats and recordings.
4. Monitoring Performance and amp; Troubleshooting
Ensure high availability low jitter low packet loss and optimal VoIP performance.
Troubleshoot SIP and RTP issues using:
sngrep
tcpdump
Wireshark
Asterisk CLI logs
Optimize codecs transcoding performance and server load.
Monitor real-time:
Queue performance
Conference performance
SLA metrics
MOS score and call quality indicators
5. Security and amp; Reliability
Implement security measures for voice and conferencing systems:
Fail2ban IP filtering SIP firewalling
Anti-fraud and intrusion prevention
Manage system upgrades patches backups and DR plans.
Deploy HA clusters for mission-critical contact center conferencing environments.
Required Skills and amp; Qualifications
35 years hands-on experience with Asterisk and amp; FreePBX (contact center conferencing).
Strong understanding of SIP RTP conferencing modules bridge applications.
Proven experience building APIs and backend services ( Python PHP Go etc.).
Hands-on experience with ARI AMI AGI and dialplan logic.
Strong Linux skills (CentOS Debian Ubuntu).
Experience configuring:
Contact center queues and routing
Conference bridges
Recordings monitoring tools
Experience with SIP trunking and carrier integrations.
Preferred / Good-to-Have Skills
Experience with large-scale conferencing platforms (Asterisk ConfBridge MeetMe or custom conferencing solutions).
Knowledge of OpenSIPS/Kamailio for scaling conferencing or call center environments.
Familiarity with WebRTC and browser-based conferencing.
Knowledge of predictive dialers like VICIdial/GoAutoDial.
Cloud deployment experience (AWS/GCP/Azure).
Familiarity with microservices and distributed architectures.
Soft Skills
Strong analytical and debugging mindset.
Ability to document systems flows and API specs clearly.
Good communication and collaboration skills.
Ability to work in fast-paced contact center or conferencing environments.
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