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<html><head><title>Re[2]: [Freeswitch-users] Don't work playback after bypass media mode.</title>
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<p>Hi, Anthony.</p>
<p><br></p>
<p>Thank's for reply.</p>
<p><br></p>
<p>You wrote 27 may 2010 Ç., 20:34:15:</p>
<p><br></p>
<p><span class=rvts13>> playback executes on line 87 of your trace.</span></p>
<p><br></p>
<p>yes, but executed by log after 30 seconds after line 84 and immediately hangup.</p>
<p>Note, duration wav file - 4 sec.</p>
<p><br></p>
<p><span class=rvts13>> If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or</span></p>
<p><span class=rvts13>the phone you are on does not support early media, </span></p>
<p><span class=rvts13><br></span></p>
<p><span class=rvts13>I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case. </span></p>
<p><span class=rvts13>If dialplans is: </span></p>
<p><span class=rvts13><extension name="fail_balance"></span></p>
<p><span class=rvts13> <condition field="${blabla}" expression="^\-1$"></span></p>
<p><span class=rvts13> <action application="pre_answer"/></span></p>
<p><span class=rvts13> <action application="sleep" data="1000"/></span></p>
<p><span class=rvts13> <action application="playback" data="elight/neg_balance.wav"/></span></p>
<p><span class=rvts13> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/></span></p>
<p><span class=rvts13> </condition></span></p>
<p><span class=rvts13> </extension></span></p>
<p><span class=rvts13>then work's fine, no problems (bypass_media=true not set). </span></p>
<p><span class=rvts13><br></span></p>
<p><span class=rvts13>> change your pre-answer before the playback to answer to verify. </span></p>
<p><br></p>
<p>If set <span class=rvts13>answer, problem disappears. But answer it's not for my case. </span></p>
<p><span class=rvts13><br></span></p>
<p><span class=rvts13>> Try adding sofia profile internal siptrace on to see the sip traffic too.</span></p>
<p><span class=rvts13><br></span></p>
<p><span class=rvts13>Call log with sip trace </span><span class=rvts22>http://pastebin.freeswitch.org/13065</span></p>
<p><br></p>
<p>Thanks again :)</p>
<p><br></p>
<p><span class=rvts10>-- </span></p>
<p><span class=rvts10>Regard's</span></p>
<p><span class=rvts10>Sergey Scheglov</span></p>
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