2010/9/17 Olivier <[email protected]> > > > 2010/9/17 Wolfgang Pichler <[email protected]> > > 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 >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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 >
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