[asterisk-users] Attended Transfer does not release channels

[Olivier]

2010/9/17 Wolfgang Pichler <wpichler at yosd.at>

> Hi all,
>
> i have the following setup
>
> PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk
> 1.6.2.9 -> SIP -> agent
>
>
> Does work quit fine - then agent does have the abibility to transfer a call
> to a third party - the agent can initiate the transfer over a web interface
> - it does generate a asterisk manager atxfer request...
>
> So agent does initiate transfer - call flow is
>
> agent -> SIP -> callcenter asterisk -> NEW call over IAX -> routing server
> -> PSTN
>
> Then agent hangs up - so that the original caller and the new call will get
> connected - and - it is working
>
> But - the call will not get released on the callcenter asterisk machine
>
> So the callflow after the transfer is
>
> Original call PSTN -> routing server -> callcenter asterisk -> routing
> server -> PSTN
>
> But it should be
>
> Original call PTN -> routing server -> PSTN
>
> I have transfer = yes and mediaonly both tested on my connection routing
> server to asterisk callcenter - does not help
>
> the iax peer beetween the both does have trunk=yes
>
> I do not get any error message (unable to transfer or something like this)
>
> I have done a full network dump of such a call - and i can see that
> asterisk callcenter does not make any attempt to directly bridge the calls -
> no TXREQ or something like that.
>
>
>
> So - why does it not try to directly bridge the both channels ?
>

see http://issues.asterisk.org/view.php?id=17999 and related bugs

>
> I am using a local channel in the middle on asterisk callcenter - with /n
> option - could this be the problem ?
>
> best regards,
> Wolfgang
>
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