Attended Transfer Problem

Scenario: Place call on trunk “A” (zap channel) to a test phone, then attempt to perform an atxfr to another line on another trunk “B”. Initially, successful, in that:
· I get the “transfer” announcement, followed by dialtone
· the original call to the test phone is placed on hold
· I get connected to the person on Trunk “B” to which I want to transfer the call

When I hang up, my expectation is that the test phone on Trunk “A” and the Trunk “B” channel will be bridged together, as per the expplanation provided at: voip-info.org/wiki-Asterisk+ … tures.conf.

Instead, the connection to Trunk “B” is disconnected and I am reconnected to the test phone.

Any help would be greatly appreciated.

On my Polycom phones, I have to hit the transfer button a second time to initiate the attended transfer. So if I’m on the phone with phone1 and I hit transfer, phone1 is put on hold and I get dialtone. I call phone2 and let them know what’s up. I then hit transfer again and I’m disconnected and phone1 and phone2 are connected together.

Maybe your phone is similar?

Thanks, Milenko. You were right, transfer works. Interestingly, I was testing using a soft phone from SJPhone. I then tried it on a Cisco phone and it worked by hanging up the call. So much for consistency.