Attended Transfer - Bridge SIP Header

Asterisk Asterisk Support
1

Is it at all possible to send a SIP Header once a bridge is created on a Attended Transferred call?

I have a customer that on invite on the WebRTC call a couple SIP Headers are passed onto their system but when a attended transfer is processed from the incoming caller side that information related to that call is required?

Regards

2

If you mean send headers after a call is answered by sending a SIP request (like a re-INVITE you trigger), then no. You can’t do that.