[Asterisk-Dev] Re: Answering a queue with an SIP UA,
then transfer to another sip UA oneway audio
fredrik at f6.nl
Tue Jun 7 05:18:33 MST 2005
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
So i suspect this is a bug in either chan_sip, or chan_agent concerning
> Help, what to do?
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> To UNSUBSCRIBE or update options visit:
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the asterisk-dev