Stuck getting Vitxi to work

Hi All,

Currently I have Vitxi going and able to ring outbound but once the call establishes I get the following:

CSeq: 4244 BYE
Reason: SIP ;cause=488; text=“Not Acceptable Here”

as per below when looking at the websocket.

BYE sip:xxxxxx:5060 SIP/2.0
Via: SIP/2.0/WSS xxxxxx;branch=z9hG4bK8870937
Max-Forwards: 69
To: sip:*72@xxxxxx;tag=b5ef4f56-6a47-4b73-973c-9196f871d8ce
From: “xxxxxx” sip:T2_1001@xxxxxx;tag=e6bh7m3kv0
Call-ID: n41or7m7a7g68aupu4gh
CSeq: 4244 BYE
BYE sip:103.110.28.193:5060 SIP/2.0
Via: SIP/2.0/WSS xxxx branch=z9hG4bK8870937
Max-Forwards: 69
To: sip:*72@xxxx;tag=b5ef4f56-6a47-4b73-973c-9196f871d8ce
From: “xxxx” sip:T2_1001@xxxx;tag=e6bh7m3kv0
Call-ID: n41or7m7a7g68aupu4gh
CSeq: 4244 BYE
Reason: SIP ;cause=488; text=“Not Acceptable Here”
REF: WEBRTC
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY,SUBSCRIBE
Supported: outbound
User-Agent: WebRTC
Content-Length: 0
REF: WEBRTC
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY,SUBSCRIBE
Supported: outbound
User-Agent: WebRTC
Content-Length: 0

Any help would be appreciated on where to look to see if I can get this working.

Hello!

Could you please share screenshots (be sure to hide any sensitive information) of the WebRTC Device Profile, Mini HTTP Settings and RTP Settings,

Best regards,

I Managed to get this going in the end, I was missing the Default WebRTC Profile on the Extension. This resolved the issue, Once I got that on there, it worked fine.

On another note tho, is it possible to set a default profile for all NEW extensions to elect this one? Or do I just edit the Default PJSIP Profile?

1 Like

Perfect!

Currently you need to manually select the desired device profile for the new extension/device.

Regards,