[Asterisk-Users] Answering a queue with an SIP UA, then transfer to another sip UA oneway audio

[fredrik chabot]

The problem is as follows

I've made a queue and i queue incoming calls in that queue.
The reception log's in as an agent to that queue and gets the calls for 
that queue so far so good.

Now I need to transfer the call. I press flash (all granstream bt101 
phones) extension, announce the call to the person and then press 
transfer. This is the normal way to transfer and works for calls who do 
not  come from the queue but now the person answering hears the caller 
but the caller does not hear the other side.

Help, what to do?

Fredrik