@kev3522 It's great that you're using a squad setup with multiple assistants for different stages of the call. The issue you're facing with the human representative not being able to hear the second assistant could be due to a few different factors. Here are some steps you can take to troubleshoot and resolve the issue: 1. **Check Volume Levels**: Ensure that the volume levels for the second assistant are set correctly and are not muted or too low. 2. **Audio Routing**: Verify that the audio from the second assistant is being correctly routed to the call once the IVR navigation is complete. 3. **PSTN Compatibility**: There might be compatibility issues with PSTN (Public Switched Telephone Network) numbers. Test the call with a PSTN number to see if the problem persists. 4. **Codec Issues**: Sometimes, the codec used for VoIP calls might not be compatible with PSTN systems. Check if the codecs used are supported by PSTN. 5. **Network Quality**: Poor network quality or latency can affect the audio transmission. Test the call quality and network conditions. 6. **Logs and Monitoring**: Review logs for any errors or warnings during the transition from the first to the second assistant. Monitoring tools can help identify if the audio stream is being sent and received properly. 7. **Test with Different Devices**: Try testing with different devices and carriers to rule out device-specific or carrier-specific issues. 8. **Contact Support**: If you've gone through these steps and still face issues, it might be best to contact support for more in-depth troubleshooting. If you need further assistance or would like to proceed with any specific checks or tests, please let me know, and I'll be happy to help you further.

@User the support bot doesn't seem to be working - would you be able to help out? thanks 🙏

I suspect I need to set up SIP trunking, so I followed https://relieved-yuzu-73a.notion.site/SIP-Outbound-Guide-997d66df35284295b7149fc10c82f3e8 but the outbound calls are failing. checked twilio error logs and I'm seeing "SIP: Invalid phone number" the "To" field looks like

<sip:1650*******@****.pstn.twilio.com>;transport=udp

when it should look like

<sip:+1650*******@****.pstn.twilio.com>;transport=udp

I'm using the javascript sdk like so to create the call:

const vapiCall = await client.calls.create({
  ...
  customer: {
    number: "+1650*******",
    numberE164CheckEnabled: false,
  }
})

despite the

+

being there in

customer.number

, it seems to always get stripped before it lands on Twilio. maybe a bug wherever Vapi makes the Twilio call?

fwiw the issue looks exactly like the one at https://discord.com/channels/1211482211119796234/1324429904757915648/

here's a call id if helpful:

f2ceb397-a9ab-41fc-b5b0-86f9120427d9

@Shubham Bajaj this is fairly urgent for us, please let me know if you could help out! 🙏

some call IDs:

7012a004-08a0-4c6e-99b6-4876772de567

,

499093a0-188c-4c4b-acd2-1720a9fb4648