I&#39;m trying to understand this:<br><br>If FS is acting as a phone in your scenario why are you sending a refer to it and not the server?<br>In most situations there is a b2bua server who routes the calls and takes all the REFER.<br>
Is this one of those PROXY only sip servers?<br><br>I think you would need to produce a full debug log of this, and if you are using some kind of proxy based setup we would need some way to easily reproduce it or visit your lab because we do not typically use anything of the sort.<br>
<br>Execute these commands and reproduce it and capture the whole log and put it on<br><a href="http://pastebin.freeswitch.org">http://pastebin.freeswitch.org</a><br><br>sofia profile internal siptrace on<br>console loglevel debug<br>
<br><br><br><br><br><div class="gmail_quote">On Wed, Apr 21, 2010 at 9:13 AM, Anthony Minessale <span dir="ltr">&lt;<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>&gt;</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;">
instead of emailing again when impatient for an answer (something we frown upon here in this busy list) <br>produce a reproducible step by step process to duplicate your issue.  We are trying to help people but we don&#39;t have the time to do the leg work for everyone who asks a question when we get hundreds of emails a day.<div>
<div></div><div class="h5"><br>
<br><br><br><div class="gmail_quote">On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <span dir="ltr">&lt;<a href="mailto:mardy@voysys.com" target="_blank">mardy@voysys.com</a>&gt;</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;">Just following up...  Does anyone have any suggestions on how to proceed with this?  I&#39;ve run out of ideas.<div><br></div><div>Thanks,</div><div><br></div><font color="#888888"><div>

-Mardy</div></font><div><div></div><div><div><br><div><div>On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:</div><br><blockquote type="cite"><div style="word-wrap: break-word;">The phones that I am using are not registered with FS.  They are registered with another proxy based PBX.  I am simply using FS as B2BUA which is also registered with the PBX.  And yes, I can successfully transfer a call to another phone with this setup.<div>

<br></div><div>To simplify things I tried the same scenario using FSComm in place of my own FS application and tried to transfer a call to FSComm with the same results.  And just in case there might be a problem specific to FSComm, I set up a clean install of FS 1.0.6 and tried transferring a call to the FS echo application with the same results.  By the way, I have no problems with blind transfers, only attended transfers.</div>

<div><br></div><div>-Mardy</div><div><br><div><div>On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:</div><br><blockquote type="cite">did you try just setting up 2 phones on plain fresh FS install, and calling them normally and transferring them around?<br>

That description is still pretty vague? What is an Event Socket application, which has nothing to do with sip and sip transfers, that&#39;s a FS protocol.<br>
<br><br><div class="gmail_quote">On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <span dir="ltr">&lt;<a href="mailto:mardy@voysys.com" target="_blank">mardy@voysys.com</a>&gt;</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;">I have two phones (Polycom) and an event_socket application, all of which are using a SIP proxy for call routing.  The first phone calls the second phone.  The second phone then attempts to transfer the call to the FS/event_socket application by first placing the call on hold and then calling the FS application, followed by a consultative transfer.  The REFER dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.  The transferred call leg appears to be answered by FS and the application receives a uuid_bridge event with the UUID of the new call leg.  The problem that I see is that the original call leg, created when the user called the FS application to announce the transfer, does not get canceled by FS and subsequently does not send the BYE back to the Polycom.  Is there something that I need to do at the event_socket application to complete the transfer?  I&#39;ve tried killing the UUID associated with the first call leg as well as issuing an &quot;answer&quot; command to the transferred call leg UUID, but no luck.<div>


<br></div><font color="#888888"><div>-Mardy</div></font><div><div></div><div><div><br><div><div><br><div><div>On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:</div><br><blockquote type="cite">but what is the client sending the REFER?<br>


<br>FS gets refer+replaces all the time, if it&#39;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">&lt;<a href="mailto:mardy@voysys.com" target="_blank">mardy@voysys.com</a>&gt;</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><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><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><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">&lt;<a href="mailto:mardy@voysys.com" target="_blank">mardy@voysys.com</a>&gt;</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&#39;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>





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