Inbound Vapi calls from Telnyx Cutting off after 4...
# support
e
Hello There! We integrated Telnyx for telephony tools. So when an agent is created on Vapi, we create a sip_uri which then conects to the respective Telnyx number. We have an occasional issue where inbound calls to vapi agents through the sip_uri keeps cutting after about 30-40 seconds on a call. I would like to understand where this issue might be coming from?
v
Based on the information I found, your call cut-off issue might be related to the
silenceTimeoutSeconds
setting. Try increasing this value in your assistant configuration:
Copy code
javascript
// Update your assistant configuration
curl -X PATCH 'https://api.vapi.ai/assistant/{ASSISTANT_ID}' \
-H 'Authorization: Bearer {YOUR_API_TOKEN}' \
-H 'Content-Type: application/json' \
-d '{
  "silenceTimeoutSeconds": 30
}'
This parameter controls how long Vapi waits during silence before ending the call. Sources: - [Vapi Documentation](https://docs.vapi.ai/api-reference/calls/create) - [Discord Discussion](https://discord.com/channels/1338586561662419055)
k
Disable SIP ALG on your network, ensure proper NAT and firewall configurations, and verify that SIP ACKs are received after the 200 OK response to prevent call drops.
e
Thank you for the response @Kings_big💫 Is this is a certain solution to the issue or something we should try out and see if it works? For further context of the situation: - Some Telnyx numbers do not have this issue - The number with this issue was working fine initially before this started. - More Call Log Details: From the vapi call logs (the transcript and recording audio), we notice only the agent's speech. We noticed that the system/agent doesn't detect that a caller is speaking and so the silenceTimeout kicks in and ends the call. This is despite the fact that the agent was giving appropriate responses to the caller's request while the call was on-going before abruptly ending. We can see a complete recording and transcript from our own streamed audio and transcription.
k
The issue likely stems from oneway audio, where the caller’s voice isn’t reaching Vapi, often due to NAT issues, SIP ALG, firewall blocks, or incorrect SIP headers. Since some Telnyx numbers work fine, compare configurations, disable SIP ALG, ensure RTP traffic is allowed, verify codec compatibility, and review SIP headers..
e
What sort issues could occur from the SIP headers? We already follow current guidelines for adding SIP headers
k
Contact Telnyx support if needed to resolve these issues..