[asterisk-users] Attended Transfer does not release channels

Wolfgang Pichler wpichler at yosd.at
Fri Sep 17 07:07:45 CDT 2010


2010/9/17 Olivier <oza_4h07 at yahoo.fr>

>
>
> 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 have taken a look at these bugs - but they don't seem to be related to my
problem - then transfer is working in my scenario - the problem is that the
call legs are not getting optimized out as it should be the case...

A calls B - B makes attended transfer to C -> B talks to C -> B hangs  up ->
asterisk should optimize out the call leg A -> B and B -> C to only A->C if
it is possible



>> 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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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
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