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

fredrik chabot 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 
invite messages...

> Help, what to do?
> Fredrik
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20050607/4602af8a/attachment.htm

More information about the asterisk-dev mailing list