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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hi,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I did a quick test against freeswitch-1.2.5.3 on CentOS 5.8, but the problem is still there.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>In the recording, the caller speaks way faster than he did.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Here's the version of freeswitch:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Here's the set of steps:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>1. In dialplan/default/resolvity.xml, add the following extension:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> <extension name="ext1"><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> <condition field="${destination_number}" expression="^0009$"><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> <action application="set" data="RECORD_STEREO=false"/><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> <action application="answer" /><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> <action application="javascript" data="test.js" /><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </condition><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </extension><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>2. Create scripts/test.js with the following content. This js file will read numbers sequentially starting from 0.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> session.execute("record_session", "/tmp/test.wav");<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> for (var i=0; i<100; i++)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> {<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> session.speak("flite", "kal", i+'');<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> }<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> session.hangup(16);<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>3. reloadxml<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>4. dial 0009 from a registered SIP client, and then repeat each number you heard.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>5. Listen to the recording, /tmp/test.wav.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Michael Collins<br><b>Sent:</b> Tuesday, December 18, 2012 8:26 PM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] voices in the recordings are out of sync<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>Latest version of FreeSWITCH has some updates that may fix this issue. I would update to 1.2.5.3 ASAP.<br>-MC<o:p></o:p></p><div><p class=MsoNormal>On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen <<a href="mailto:yungwei@resolvity.com" target="_blank">yungwei@resolvity.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Hi,<br><br>I found one issue that voices are always out of sync in the recordings.<br>I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed from yum.<br>I am having trouble installing the latest version from source due to an error: Autoconf version 2.62 or higher is required.<br>It would be nice if someone can reproduce this issue against HEAD. Thanks.<br><br>Here're the steps to reproduce it. The idea is to call a phone number and then bridge to another phone number while the entire session is being recorded.<br>1. In dialplan/public.xml, make sure you have a dialplan to handle any 10 digit phone numbers.<br> <extension name="public_extensions"><br> <condition field="destination_number" expression="^\d{10}$"><br> <action application="transfer" data="main XML default"/><br> </condition><br> </extension><br><br>2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context.<br> <extension name="test"><br> <condition field="${destination_number}" expression="^main$"><br> <action application="set" data="RECORD_STEREO=false"/><br> <action application="answer" /><br> <action application="record_session" data="/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/><br> <action application="bridge" data="{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890"/><br> </condition><br> </extension><br><br>3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call.<br> <include><br> <gateway name="gw1"><br> <param name="username" value=""/><br> <param name="password" value=""/><br> <param name="realm" value=""/><br> <param name="from-domain" value=""/><br> <param name="extension" value=""/><br> <param name="expire-seconds" value="60"/><br> <param name="register" value="false"/><br> <param name="retry-seconds" value="60"/><br> </gateway><br> </include><br><br>4. make a call to one of the allowed 10-digit phone numbers in your environment.<br>5. Once the call is answered, the caller shall start to count from 1 to 60 with some pause after each number.<br>6. The callee shall repeat each number he/she heard from the caller.<br>7. You should be able to hear that 2 voices in the recoridng (/tmp/rec.wav) are out of sync.<br><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br><a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br><a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br><br>Official FreeSWITCH Sites<br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br><a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><o:p></o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'><br><br clear=all><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br><a href="http://www.OSTAG.org" target="_blank">http://www.OSTAG.org</a><br><br><o:p></o:p></p></div></body></html>