Fix FreePBX and ElevenLabs SIP Integration/Asterisk
Budget: $30 – $250 USD
We are looking for an experienced VoIP engineer to fix and finalize a SIP integration between FreePBX and ElevenLabs AI, with Twilio used for external PSTN calls.
Please do not apply unless you have deep, hands-on experience with Asterisk 20, FreePBX 17, and PJSIP.
Objectives (Must Be Achieved)
Scenario 1 – Internal Test
• Extension 1001 calls ElevenLabs AI via SIP
• ElevenLabs processes the call
• ElevenLabs transfers the call back to extension 1002
Scenario 2 – External Call
• Call arrives from a Twilio number
• Routed to FreePBX
• Forwarded to ElevenLabs AI
• AI transfers the call to a live PBX extension
Both scenarios must be fully working and tested
Current Environment
• PBX: FreePBX 17
• Asterisk: 20.x
• OS: Ubuntu 22.04
• Hosting: AWS EC2
• Signaling: PJSIP only
• Transport tested: UDP / TCP (TLS optional)
External Services
• ElevenLabs SIP Trunk (Agent Voice / AVR)Current Issues (What Needs Fixing)
• ElevenLabs SIP trunk shows:
o Rejected
o No auth objects matching realm
• Calls connect but no audio
• “All circuits are busy” after dialing
• Unclear authentication behavior between PBX and ElevenLabs
• Suspected RTP / NAT / media path issues
• Need correct final configuration on:
o FreePBX (GUI and Asterisk level)
o ElevenLabs dashboard
Required Skills (DO NOT APPLY WITHOUT THESE)
• Experience with Asterisk and FreePBX
• Deep PJSIP knowledge (endpoint, AOR, auth, identify)
• SIP trunking with cloud AI / voice bots
• RTP, NAT, and audio troubleshooting
• AWS networking and security groups
• Ability to read SIP traces, tcpdump, and pcap files
• Twilio SIP and PSTN routing (strong bonus)
Deliverables
• Fully working PBX,ElevenLabs SIP integration
• Both scenarios tested and confirmed
• Final documentation covering:
o SIP trunk configuration
o Dialplan and call flow logic
o ElevenLabs settings
Optional (Nice to Have)
• Security hardening and best-practice recommendations
Time
• Estimated time: 3–6 hours
Important Notes
• This is not a trial-and-error task
• You must understand how ElevenLabs SIP trunking actually works
• You must be comfortable working live on the server (SSH and FreePBX GUI)
How to Apply (Required)
Please include all of the following:
1. Previous Asterisk / FreePBX SIP trunk projects
2. Experience with AI voice agents or SIP bots
3. Confirmation that you can start immediately
Applications without this information will be ignored
Closing
Please mention the word (ElevenLabs) at the beginning of your BID
We are looking for one expert who knows exactly what they are doing, not multiple attempts.
If successful, this can lead to ongoing VoIP and AI telephony work.
Please do not apply unless you have deep, hands-on experience with Asterisk 20, FreePBX 17, and PJSIP.
Objectives (Must Be Achieved)
Scenario 1 – Internal Test
• Extension 1001 calls ElevenLabs AI via SIP
• ElevenLabs processes the call
• ElevenLabs transfers the call back to extension 1002
Scenario 2 – External Call
• Call arrives from a Twilio number
• Routed to FreePBX
• Forwarded to ElevenLabs AI
• AI transfers the call to a live PBX extension
Both scenarios must be fully working and tested
Current Environment
• PBX: FreePBX 17
• Asterisk: 20.x
• OS: Ubuntu 22.04
• Hosting: AWS EC2
• Signaling: PJSIP only
• Transport tested: UDP / TCP (TLS optional)
External Services
• ElevenLabs SIP Trunk (Agent Voice / AVR)Current Issues (What Needs Fixing)
• ElevenLabs SIP trunk shows:
o Rejected
o No auth objects matching realm
• Calls connect but no audio
• “All circuits are busy” after dialing
• Unclear authentication behavior between PBX and ElevenLabs
• Suspected RTP / NAT / media path issues
• Need correct final configuration on:
o FreePBX (GUI and Asterisk level)
o ElevenLabs dashboard
Required Skills (DO NOT APPLY WITHOUT THESE)
• Experience with Asterisk and FreePBX
• Deep PJSIP knowledge (endpoint, AOR, auth, identify)
• SIP trunking with cloud AI / voice bots
• RTP, NAT, and audio troubleshooting
• AWS networking and security groups
• Ability to read SIP traces, tcpdump, and pcap files
• Twilio SIP and PSTN routing (strong bonus)
Deliverables
• Fully working PBX,ElevenLabs SIP integration
• Both scenarios tested and confirmed
• Final documentation covering:
o SIP trunk configuration
o Dialplan and call flow logic
o ElevenLabs settings
Optional (Nice to Have)
• Security hardening and best-practice recommendations
Time
• Estimated time: 3–6 hours
Important Notes
• This is not a trial-and-error task
• You must understand how ElevenLabs SIP trunking actually works
• You must be comfortable working live on the server (SSH and FreePBX GUI)
How to Apply (Required)
Please include all of the following:
1. Previous Asterisk / FreePBX SIP trunk projects
2. Experience with AI voice agents or SIP bots
3. Confirmation that you can start immediately
Applications without this information will be ignored
Closing
Please mention the word (ElevenLabs) at the beginning of your BID
We are looking for one expert who knows exactly what they are doing, not multiple attempts.
If successful, this can lead to ongoing VoIP and AI telephony work.