<div class="gmail_quote">2010/9/17 Olivier <span dir="ltr"><<a href="mailto:oza_4h07@yahoo.fr">oza_4h07@yahoo.fr</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<br><br><div class="gmail_quote">2010/9/17 Wolfgang Pichler <span dir="ltr"><<a href="mailto:wpichler@yosd.at" target="_blank">wpichler@yosd.at</a>></span><div><div></div><div class="h5"><br><blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">
Hi all,<div><br></div><div>i have the following setup</div><div><br></div><div>PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent</div><div><br></div><div><br>
</div>
<div>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...</div><div>
<br></div><div>So agent does initiate transfer - call flow is</div><div><br></div><div>agent -> SIP -> callcenter asterisk -> NEW call over IAX -> routing server -> PSTN</div><div><br></div><div>Then agent hangs up - so that the original caller and the new call will get connected - and - it is working</div>
<div><br></div><div>But - the call will not get released on the callcenter asterisk machine</div><div><br></div><div>So the callflow after the transfer is</div><div><br></div><div>Original call PSTN -> routing server -> callcenter asterisk -> routing server -> PSTN</div>
<div><br></div><div>But it should be</div><div><br></div><div>Original call PTN -> routing server -> PSTN</div><div><br></div><div>I have transfer = yes and mediaonly both tested on my connection routing server to asterisk callcenter - does not help</div>
<div><br></div><div>the iax peer beetween the both does have trunk=yes</div><div><br></div><div>I do not get any error message (unable to transfer or something like this)</div><div><br></div><div>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.</div>
<div><br></div><div><br></div><div><br></div><div>So - why does it not try to directly bridge the both channels ?</div></blockquote></div></div><div><br>see <a href="http://issues.asterisk.org/view.php?id=17999" target="_blank">http://issues.asterisk.org/view.php?id=17999</a> and related bugs<br>
</div></div></blockquote><div>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...</div>
<div><br></div><div>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</div><div><br></div>
<div><br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div class="gmail_quote"><div>
</div><blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex"><div class="im"><div><br></div><div>I am using a local channel in the middle on asterisk callcenter - with /n option - could this be the problem ?</div>
<div><br></div><div>best regards,</div><div>Wolfgang</div>
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