Handling Attended Transfer


I have an asterisk server 13 and I know that AMI events can detect a attended transfer but I want somehow without an external service to detect it just to flag it and maybe write channel variables after the bridge.

We are using PortSIP lib and IP phones.

Can you give me some ideas how can I handle attended transfer?

features.conf or native SIP?

Native SIP. I need to somehow know that the call was mixed. Is there any way I can detect that without AMI event?

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