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Genesys cloud Sip Trunk Integration
# support
l
I’ve set up a Sip Trunk for the Vapi integration but i keep getting a error: Call.in progress.error sip outbound call failed to connect Any help would be appreciated. I am using Genesys cloud.
v
Action Step: Make sure you have allowlisted the following Vapi IP addresses in your Genesys Cloud SIP trunk/network settings, as missing this step often causes outbound call failures:
Copy code
44.229.228.186/32
44.238.177.138/32
After updating your firewall or network rules, try your outbound call again. For more details, see the [Vapi SIP trunk documentation](https://docs.vapi.ai/advanced/sip/sip-trunk). Source: - [Vapi SIP Trunk Documentation](https://docs.vapi.ai/advanced/sip/sip-trunk)
l
I have those as part of the allow list
j
I can help you troubleshoot this. The “Call in progress – SIP outbound call failed to connect” often points to either SIP trunk configuration issues or network/firewall blocks. Can you confirm if your trunk is fully registered in Genesys Cloud and that the required SIP ports are open end-to-end? Also, are you seeing this with all outbound calls or specific numbers?. @Luke
l
Hi @Jarrah, I appreciate your help. I am configuring this for the first time so I’ve only tried to set this up for a single phone number. If it works for you, I will post the configs I have on my end (as best anonymized as possible, and maybe you can give advice.
In Vapi, I created a new sip trunk with 3 separate gateways. I looked on Genesys’s public BYOC Cloud SIP Ip Address and added the 3 ingress IP addresses for my region. 52.129.98.228 52.129.98.229 52.129.98.230 I left the port (5060), Netmask(32), protocol (UDP) as the defaults.
For the authentication side I’m not too sure what I’m supposed to do. I tried to match the Genesys config settings as close to Vapis for auth. So I’ll explain what I did in Genesys. I went to their external trunks and created a “Generic BYOC Carrier” trunk. For the settings I have Protocol:UDP Termination Identifier: {unique identifier} Digest authentication: Enabled Realm : {Realm} User name: {username} Password: {password} Sip access control, allow the following addresses: 44.238.177.138/32 and 44.229.228.186/32 all other settings are defaulted
In Vapis Authentication I used Username: {Genesys username for external trunk} Password: {Genesys password for external trunk} Use Sip Registration: true Domain: {Genesys unique terminator for external trunk} Username : {Genesys user name for external trunk} Realm: {Genesys realm for external trunk}
Sorry to clarify for the domain i used the FQDN provided domain. {unique identifier}.byoc.use1.pure.cloud
c
Hi @Luke, could you provide the Call ID of the call which ended with Call.in-progress.error? I can review the logs and see what issue you are facing?
l
019bfc3e-3cc1-711d-bd65-7b1c4273b96c
c
Hi, Thank you for providing the call ID, I reviewed the call and it looks like your SIP provider returned with an 404 Not Found error. and from looking into the SIP logs, it looks like you have kept the To header as 
<phone_number>@APTIVA
. APTIVA doesnt look like a proper domain name. Could you confirm whether you wrongly kept the destination domain?
l
It should be @aptiva.byoc.use1.pure.cloud
Do you know which settings I have to change to configure this
If it would be best to hop in a call together I can do anytime
c
Hi, Please delete the existing SIP trunking setup you have and create a new one. More information about how to create an SIP trunk here: https://docs.vapi.ai/advanced/sip/sip-trunk#setup-process. Make sure to enter the correct URL in the gateway (aptiva.byoc.use1.pure.cloud)
l
Is there any documentation on how to set it up through the dashboard?
Or is there any way I can try to diagnose this issue, like how you saw what To header I sent
c
You can do it from the dashboard, Go to Vapi dashboard -> integrations -> SIP Trunk the settings you will see will be similar to the API request body. However I recommend using the API, since it will provide more information about the responses. Regarding how I got the header, I got it from backend logs which are not accessible by the end user.
l
Ok I will try the api
I got a 403 forbidden error
I don’t know what my next steps would be
c
Could you share the API response with me?
l
Copy code
{
  "subscriptionLimits": {
    "concurrencyBlocked": false,
    "concurrencyLimit": 10,
    "remainingConcurrentCalls": 9
  },
  "id": "019c0b48-412f-7665-b99b-0f7c2f128d0f",
  "assistantId": "65406b85-3a6f-4963-8b2a-fd563ec19525",
  "phoneNumberId": "ff27c810-cf10-40df-8ef0-9a1a8d8e8649",
  "type": "outboundPhoneCall",
  "createdAt": "2026-01-29T19:43:30.607Z",
  "updatedAt": "2026-01-29T19:43:30.607Z",
  "orgId": "8b065818-36a5-49f7-b7da-23ff37706a5f",
  "cost": 0,
  "customer": {
    "number": "+19544384070",
    "numberE164CheckEnabled": false
  },
  "status": "queued",
  "phoneCallProvider": "vapi",
  "monitor": {
    "listenUrl": "wss://phone-call-websocket.aws-us-west-2-backend-production2.vapi.ai/019c0b48-412f-7665-b99b-0f7c2f128d0f/listen",
    "controlUrl": "https://phone-call-websocket.aws-us-west-2-backend-production2.vapi.ai/019c0b48-412f-7665-b99b-0f7c2f128d0f/control"
  },
  "transport": {
    "conversationType": "voice",
    "provider": "vapi.sip"
  }
}
c
The response you shared had this call ID 
a17d3e3f-77ed-123f-61b0-0a8f09508ca9
and it returned with 404 error, the same error which i mentioned earlier. You are not setting up the domain correctly. Please delete your existing SIP trunk integration on Vapi and follow this guide to setup a new one here: •

https://youtu.be/_wo5wokt3dI▾

• [https://docs.vapi.ai/advanced/sip/sip-trunk](https://docs.vapi.ai/advanced/sip/sip-trunk#setup-process)
l
Do you know if its possible to set up VAPI Sip Trunk with a Genesys BYOC Carrier external trunk?
Is there any way we can hop on a quick call or I can get in touch with someone for assistance?
c
Hi @Luke, You can follow the same tutorial for Genesys, most of the information provided should be similar for many SIP providers. However we are not able to accommodate calls at the moment, and there are a lot of SIP providers, we are not able to provide support for all of them. The ones we provide are listed in the documentation. I hope you understand.
l
nevermind I solved it
I'll add a soltuion so if other users run into the issue they can check this
g
Hi @Luke, The providers I work with will change from Vicidial to Genesys. Any advice to avoid unexpected behaviors on both sides?
l
In Genesys create an external trunk (Type:Generic BYOC PBX)
if you need help on exact configs just msg me and I can walk you through it
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