[asterisk-users] Attended Transfer does not release channels

[Wolfgang Pichler]

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 ?

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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