I figured that this is due to a bug in my code (bad event handling) by creating a lua script that performs exactly the same steps as the java app.<div><br></div><div>Thanks for your help,</div><div>George</div><div><br><div class="gmail_quote">
2010/12/28 George Niculae <span dir="ltr"><<a href="mailto:george.niculae79@gmail.com">george.niculae79@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hi Michael, <div><br></div><div>is the provided trace OK or should I collect new logs? </div><div><div><br></div><div>Thanks,</div><div>George<br><br><div class="gmail_quote">2010/12/23 George Niculae <span dir="ltr"><<a href="mailto:george.niculae79@gmail.com" target="_blank">george.niculae79@gmail.com</a>></span><div>
<div></div><div class="h5"><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Here it is: <a href="http://pastebin.freeswitch.org/14873" target="_blank">http://pastebin.freeswitch.org/14873</a><div>
<br></div><div><a href="http://pastebin.freeswitch.org/14873" target="_blank"></a>Thanks,</div>
<div>George<br><br><div class="gmail_quote">2010/12/23 Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span><div><div></div><div><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Try turning on the siptrace as well so we can see the sip traffic:<div><br></div><div>sofia profile internal siptrace on</div><div><br></div><div>Then do another test & pastebin the debug output.</div><div>-MC<div><div>
</div><div><br><br><div class="gmail_quote">
On Thu, Dec 23, 2010 at 4:36 AM, George Niculae <span dir="ltr"><<a href="mailto:george@ezuce.com" target="_blank">george@ezuce.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Michael,<br>
<br>
the commands are written on socket using PrintWriter.printf() and in<br>
this case is something like:<br>
api uuid_deflect f5539b24-0e8e-11e0-9a0e-c37fe40448c1 <a href="mailto:sip%3A101@dizzy.dizzysip.ro" target="_blank">sip:101@dizzy.dizzysip.ro</a><br>
Please see here all commands sent (prefixed with FSES::cmd):<br>
<a href="http://pastebin.freeswitch.org/14868" target="_blank">http://pastebin.freeswitch.org/14868</a> , uuid deflect at line 27<br>
New console output (for correlating uuid's if needed):<br>
<a href="http://pastebin.freeswitch.org/14867" target="_blank">http://pastebin.freeswitch.org/14867</a><br>
<br>
Thanks,<br>
<font color="#888888">George<br>
</font><div><div></div><div><br>
On Thu, Dec 23, 2010 at 2:07 AM, Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>> wrote:<br>
> Please pastebin the code that performs the uuid_deflect so that we can see<br>
> what you are doing to produce this symptom.<br>
> -MC<br>
><br>
> On Wed, Dec 22, 2010 at 8:35 AM, George Niculae <<a href="mailto:george@ezuce.com" target="_blank">george@ezuce.com</a>> wrote:<br>
>><br>
>> Hi All,<br>
>><br>
>> I am working on an IVR application based on FS (running FreeSWITCH<br>
>> Version 1.0.head (git-43393f2 2010-12-15 20-59-42 -0600) where the<br>
>> following scenario fails:<br>
>> user 201 calls to 100 (autoattendant), hears menu then press # to<br>
>> transfer to voicemail (101), but the call is dropped (transfer is made<br>
>> using uuid_deflect api command)<br>
>> Dialplan extension configured like:<br>
>><br>
>> <extension name="IVR"><br>
>> <condition field="destination_number" expression="^IVR${dollar}"><br>
>> <action application="socket" data="<a href="http://192.168.0.114:8084" target="_blank">192.168.0.114:8084</a> async full"/><br>
>> </condition><br>
>> </extension><br>
>><br>
>> Actions taken are:<br>
>> - when call arrives to extension 100 call is bridged<br>
>> (hangup_after_bridge=true)<br>
>> - answer the call, autoattendant menu is played and DTMF collected<br>
>> - when # pressed, call is transfered to 101 using uuid_deflect<br>
>> - call arrives to voicemail extension and is again bridged<br>
>> - call is answered - at this point in time the initial bridge hangs up<br>
>> and the whole call is dropped<br>
>> Please see console output <a href="http://pastebin.freeswitch.org/14855" target="_blank">http://pastebin.freeswitch.org/14855</a><br>
>><br>
>> When debugging the application, If I keep the first channel connected<br>
>> transfer works just fine without dropping the call.<br>
>> Pretty sure I'm missing something here, any suggestion highly appreciated<br>
>><br>
>> Thanks,<br>
>> George<br>
>><br>
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