but what is the client sending the REFER?<br><br>FS gets refer+replaces all the time, if it's the one where the dest is on another box (aka the nightmare xfer that you should see references to in the debug log if so) then it will not complete until that far end call is answered.<br>
<br>FS handles this scenerio for us hundreds of times a day using a wide range of sip devices so perhaps<br>your UA has an interop problem.<br><br><br><div class="gmail_quote">On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <span dir="ltr"><<a href="mailto:mardy@voysys.com">mardy@voysys.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div style="word-wrap: break-word;"><br><div><div class="im"><div>On Apr 19, 2010, at 5:12 PM, Joćo Mesquita wrote:</div>
<br><blockquote type="cite">uuid_simplify will issue the refer...</blockquote><div><br></div></div>I looked at uuid_simplify and if I understand it correctly it is for use when one wants to act as the transfer controller. In my case, FS is the transfer destination. Another phone has already generated the refer and FS has been sent an invite with replaces.</div>
<div><div class="im"><br><blockquote type="cite"><div><br></div><div>May I ask what application you are developing?</div></blockquote><div><br></div></div><div>An ACD.</div><div><div></div><div class="h5"><br><blockquote type="cite">
<div><br></div><div>Regards,<br clear="all">Joćo Mesquita<br>FSComm developer<br>
<br><br><div class="gmail_quote">On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <span dir="ltr"><<a href="mailto:mardy@voysys.com" target="_blank">mardy@voysys.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm. (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.) What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller. From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active. I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call. Is there a specific action that the application must take in order for the transfer to complete?<br>
<br>
-Mardy<br>
<br>
<br>
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