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

[fredrik chabot]

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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