Try turning on the siptrace as well so we can see the sip traffic: sofia profile internal siptrace on Then do another test & pastebin the debug output. -MC On Thu, Dec 23, 2010 at 4:36 AM, George Niculae <george@ezuce.com> wrote:
Michael,
the commands are written on socket using PrintWriter.printf() and in this case is something like: api uuid_deflect f5539b24-0e8e-11e0-9a0e-c37fe40448c1 sip:101@dizzy.dizzysip.ro <sip%3A101@dizzy.dizzysip.ro> Please see here all commands sent (prefixed with FSES::cmd): http://pastebin.freeswitch.org/14868 , uuid deflect at line 27 New console output (for correlating uuid's if needed): http://pastebin.freeswitch.org/14867
Thanks, George
On Thu, Dec 23, 2010 at 2:07 AM, Michael Collins <msc@freeswitch.org> wrote:
Please pastebin the code that performs the uuid_deflect so that we can see what you are doing to produce this symptom. -MC
On Wed, Dec 22, 2010 at 8:35 AM, George Niculae <george@ezuce.com> wrote:
Hi All,
I am working on an IVR application based on FS (running FreeSWITCH Version 1.0.head (git-43393f2 2010-12-15 20-59-42 -0600) where the following scenario fails: user 201 calls to 100 (autoattendant), hears menu then press # to transfer to voicemail (101), but the call is dropped (transfer is made using uuid_deflect api command) Dialplan extension configured like:
<extension name="IVR"> <condition field="destination_number" expression="^IVR${dollar}"> <action application="socket" data="192.168.0.114:8084 async
full"/>
</condition> </extension>
Actions taken are: - when call arrives to extension 100 call is bridged (hangup_after_bridge=true) - answer the call, autoattendant menu is played and DTMF collected - when # pressed, call is transfered to 101 using uuid_deflect - call arrives to voicemail extension and is again bridged - call is answered - at this point in time the initial bridge hangs up and the whole call is dropped Please see console output http://pastebin.freeswitch.org/14855
When debugging the application, If I keep the first channel connected transfer works just fine without dropping the call. Pretty sure I'm missing something here, any suggestion highly appreciated
Thanks, George
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org