Implementing AI Assistant with Call Transfer Capab...
# supportr
rpv
04/10/2025, 1:27 PMI'm working on an AI assistant with capabilities through webhooks, specifically for a FreePBX/Asterisk telephony system. The assistant would be accessible via a dedicated phone number through SIP trunking.
Proposed Call Transfer Flow:
1. Human caller connects with AI assistant
2. When caller requests a human specialist, the AI triggers a
call_transferv
Vapi Support Bot
04/10/2025, 1:30 PMHi there! đź‘‹ I'd be happy to help with your questions about implementing an AI assistant with call transfer capabilities in FreePBX/Asterisk using Vapi.
I can address all of your key questions based on the information I've found about Vapi's capabilities and integration with FreePBX/Asterisk systems.
## Is this implementation approach feasible with Vapi?
Yes, your proposed call transfer flow is feasible with Vapi! According to [Vapi's documentation](https://docs.vapi.ai/call-forwarding), Vapi supports call transfers using the
transferCalltransfer_callasync: falsetool_delayVapi Support Bot
04/10/2025, 1:30 PMVapi's behavior during call transfers depends on several factors:
1. After the transfer is complete, Vapi can provide confirmation that the call has been successfully transferred.
2. When your server receives the transfer call destination request, you should keep that HTTP connection open while checking the representative.
3. Vapi doesn't set any timeouts—as soon as it receives a status callback from Twilio, it updates the status.
However, there are some potential issues to be aware of:
- Vapi might end the call when the assistant disconnects
- There could be immediate hangup after a "Success" webhook result due to configuration issues
## How does Vapi integrate with FreePBX/Asterisk systems?
Vapi integrates with FreePBX/Asterisk primarily through SIP trunking. Here's how to set it up:
1. Set up a SIP trunk in Asterisk that points to Vapi's SIP endpoint
2. Configure your Asterisk server to connect to Vapi as a SIP trunk
3. Set up the dial plan in Asterisk to route outbound calls through the Vapi SIP trunk
4. For inbound calls, configure a SIP URI in your Vapi account and point it to your Asterisk server
For call transfers, Vapi utilizes SIP REFER messages, so ensure that your FreePBX/Asterisk system is configured to handle these messages correctly. If your SIP endpoint requires authentication, double-check that the credentials provided to Vapi are correct.
## Maintaining the AI's conversational context during the call transfer process
To maintain the AI's conversational context during call transfers, you can:
1. **Use SIP headers to pass context**: Pass context from your servers to Vapi and back using SIP headers when forwarding a call.
```javascript
const transferDestination = {
type: 'sip',
sipUri: 'sip:genesys@your-domain.com',
transferPlan: {
mode: 'blind-transfer-add-summary-to-sip-header',
summaryPlan: {
// Configure how the summary should be generated
}
},
sipHeaders: {
'X-Caller-Intent': 'policy_inquiry',
Vapi Support Bot
04/10/2025, 1:30 PM'X-Caller-Details': JSON.stringify({
name: customer.name,
policy: customer.policyNumber
})
}
};
```
2. **Utilize warm transfers**: Vapi supports warm transfers, where context about the call is provided to the receiving agent.
3. **Configure `transferMode`**: Set the
transferModerolling-historyVapi Support Bot
04/10/2025, 1:30 PM- [Discord thread about maintaining context](https://discord.com/channels/1211482211119796234/1211483291191083018/threads/1339303957675245669)
r
rpv
04/10/2025, 1:49 PMThank you. The response incorrectly assumes I'm using Twilio, but I explicitly mentioned that I have a FreePBX/Asterisk system.
rpv
04/10/2025, 1:50 PMI'm familiar with that example using Twilio because I've read the docs, but that doesn't work for me.
k
Kyle Brunker
04/11/2025, 7:12 PMHey RPV, I will be answering your key questions in the following order:
1. Implementation approach
2. During the on-hold period
3. To integrate your FreePBX or Asterisk system with VAPI
Yes, the implementation approach is feasible with VAPI - no doubt in that part. For the during the on-hold period, as of now, VAPI doesn't have any native logic or feature to check if a transfer is successful or unsuccessful. You have to handle it by yourself on the telephony side. As of now, what we do is we send a REFER request followed by a BYE request where we just do the transfer from our side without checking if there is someone to pick the call or not. In case webhook result comes at max, we can wait for a dynamic destination, but we won't be waiting if someone connects to the call on the other side or not. To integrate your FreePBX or Asterisk system with VAPI, the process is similar as I already mentioned. You went through the docs, so you have to follow the SIP trunking guide and get things done. For your information, when you make inbound calls, VAPI requires authorization, and at this time, you are supposed to return a username which has to be your SIP username registered with VAPI and then a password which can be anything except being empty white spaces.
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