fredrik chabot wrote:
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.
To update myself; I have included in the [general] section of sip.conf
the statement "canreinvite=NO" and now I do have twoway audio.
All phones and the asterisk server I have used to test this are on the
same subnet.
So i suspect this is a bug in either chan_sip, or chan_agent concerning
invite messages...
Help, what to do?
Fredrik
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