Wolfgang Pichler
2010-Sep-17 09:31 UTC
[asterisk-users] Attended Transfer does not release channels
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100917/a618438f/attachment.htm
Olivier
2010-Sep-17 11:01 UTC
[asterisk-users] Attended Transfer does not release channels
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 > > -- > _____________________________________________________________________ > -- 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 > > asterisk-users mailing list > 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-users/attachments/20100917/5fbbc98a/attachment.htm