Attended transfer issue?

Hi! I’ve got a few questions about busy tones in the attended transfer.

I’m running Asterisk 11.5 with IP phones.

1.) User ‘A’ calls ‘B’.

2.) ‘B’ attd-transfers to ‘C’. But ‘C’ is already speaking on phone

3.) Asterisk, then, connects again A to B without any notice to ‘B’.
‘B’ wonders why is he talkign to ‘A’… B thinks that attended-transfer doesn’t work.

How should I treat the busy part in dialplan when attended-transfering? I would like ‘B’ to hear a busy tone if C is already speaking to someone else.

Thanks.

Are you using features transfers or SIP transfers? If you are using SIP transfers,the phone may be doing all this. Note that some phones do SIP blind transfers as though they were attended ones.

I’m using attended transfer with dtmf ( features.conf)

I manage busy tone in dialplan with a macro:

[code][macro-estadohangup]
exten => s,1,GotoIf($[${ISNULL(${DIALSTATUS})}]?2:rc-${DIALSTATUS},1)
exten => s,2,Set(RC=${IF($[${ISNULL(${HANGUPCAUSE})}]?0:${HANGUPCAUSE})})
exten => s,3,Goto(rc-${RC},1)
exten => s,4,Hangup(${RC})
exten => i,1,Set(RC=0)
exten => i,2,Goto(s,4)

; Convert DIALSTATUS to HANGUPCAUSE
exten => rc-BUSY,1,Set(RC=17)
exten => rc-BUSY,n,Answer()
exten => rc-BUSY,n,Playtones(busy)
exten => rc-BUSY,n,Goto(s,4)

exten => rc-ANSWER,1,…
exten => rc-CANCEL,1,…
exten => rc-CONGESTION,1,…
exten => rc-CHANUNAVAIL,1,…

[/code]

But this macro never gets called when users execute featured attended-transfer.

How can I handle busy tone? Is there anything wroing in my setup?

How does the macro get called when you are not doing a “blonde” transfer?

(I suspect there is no end user configurable way of doing what you want to do.)

OK, I didn’t know that this behaviour was the expected one. :smile:

Thanks.