<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 14 (filtered medium)"><title>Re: [Freeswitch-users] voices in the recordings are out of sync</title><style><!--
/* Font Definitions */
@font-face
        {font-family:Calibri;
        panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
        {font-family:Tahoma;
        panose-1:2 11 6 4 3 5 4 4 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
        {margin:0cm;
        margin-bottom:.0001pt;
        font-size:12.0pt;
        font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
        {mso-style-priority:99;
        color:blue;
        text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
        {mso-style-priority:99;
        color:purple;
        text-decoration:underline;}
span.EmailStyle17
        {mso-style-type:personal-reply;
        font-family:"Calibri","sans-serif";
        color:#1F497D;}
.MsoChpDefault
        {mso-style-type:export-only;
        font-size:10.0pt;}
@page WordSection1
        {size:612.0pt 792.0pt;
        margin:72.0pt 72.0pt 72.0pt 72.0pt;}
div.WordSection1
        {page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</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'>Did you try other voices?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Are you sure you have built the latest version of flite?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Download and build flite again.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I&#8217;m pretty sure this was fixed some time ago.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><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>Ken Rice<br><b>Sent:</b> Monday, January 28, 2013 23:58<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></div><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:11.0pt;font-family:"Courier New"'>Ok... You should be testing against MASTER... If you are not testing again Master It is not going to get fixed... All patches go there first... If you can duplicate there, then open a ticket. This is the only way these sort of things get fixed<br><br><br><br>On 1/28/13 4:43 PM, &quot;Yungwei Chen&quot; &lt;<a href="yungwei@resolvity.com">yungwei@resolvity.com</a>&gt; wrote:</span><o:p></o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Tested against the following version today, and the problem is still there. <br>FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 2013-01-04 15:45:59Z)<br>&nbsp;<br></span><span style='font-size:11.0pt;font-family:"Courier New"'><br></span><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"'> <a href="freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Yungwei Chen<br><b>Sent:</b> Friday, January 25, 2013 5:07 PM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] voices in the recordings are out of sync<br></span><br><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hi,<br>&nbsp;<br>I did a quick test against freeswitch-1.2.5.3 on CentOS 5.8, but the problem is still there.<br>In the recording, the caller speaks way faster than he did.<br>&nbsp;<br>Here's the version of freeswitch:<br>FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z)<br>&nbsp;<br>Here's the set of steps:<br>1. In dialplan/default/resolvity.xml, add the following extension:<br>&nbsp;&nbsp;&lt;extension name=&quot;ext1&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;condition field=&quot;${destination_number}&quot; expression=&quot;^0009$&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;set&quot; data=&quot;RECORD_STEREO=false&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;answer&quot; /&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;javascript&quot; data=&quot;test.js&quot; /&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;/condition&gt;<br>&nbsp;&nbsp;&lt;/extension&gt;<br>&nbsp;<br>2. Create scripts/test.js with the following content. This js file will read numbers sequentially starting from 0.<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;session.execute(&quot;record_session&quot;, &quot;/tmp/test.wav&quot;);<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;for (var i=0; i&lt;100; i++)<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;{<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;session.speak(&quot;flite&quot;, &quot;kal&quot;, i+'');<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;}<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;session.hangup(16);<br>&nbsp;<br>3. reloadxml<br>4. dial 0009 from a registered SIP client, and then repeat each number you heard.<br>5. Listen to the recording, /tmp/test.wav.<br>&nbsp;<br>&nbsp;<br></span><span style='font-size:11.0pt;font-family:"Courier New"'><br></span><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"'> <a href="freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>] <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<br></span><br>Latest version of FreeSWITCH has some updates that may fix this issue. I would update to 1.2.5.3 ASAP.<br>-MC<br><span style='font-size:11.0pt;font-family:"Courier New"'><br></span>On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen &lt;<a href="yungwei@resolvity.com">yungwei@resolvity.com</a>&gt; wrote:<br>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>&nbsp;&nbsp;&nbsp;&nbsp;&lt;extension name=&quot;public_extensions&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;condition field=&quot;destination_number&quot; expression=&quot;^\d{10}$&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;transfer&quot; data=&quot;main XML default&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;/condition&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&lt;/extension&gt;<br><br>2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context.<br>&nbsp;&nbsp;&lt;extension name=&quot;test&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;condition field=&quot;${destination_number}&quot; expression=&quot;^main$&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;set&quot; data=&quot;RECORD_STEREO=false&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;answer&quot; /&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;record_session&quot; data=&quot;/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;bridge&quot; data=&quot;{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;/condition&gt;<br>&nbsp;&nbsp;&lt;/extension&gt;<br><br>3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call.<br>&nbsp;&nbsp;&nbsp;&nbsp;&lt;include&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;gateway name=&quot;gw1&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;param name=&quot;username&quot; value=&quot;&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;param name=&quot;password&quot; value=&quot;&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;param name=&quot;realm&quot; value=&quot;&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;param name=&quot;from-domain&quot; value=&quot;&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;param name=&quot;extension&quot; value=&quot;&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;param name=&quot;expire-seconds&quot; value=&quot;60&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;param name=&quot;register&quot; value=&quot;false&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;param name=&quot;retry-seconds&quot; value=&quot;60&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;/gateway&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&lt;/include&gt;<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="consulting@freeswitch.org">consulting@freeswitch.org</a><br><a href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a><br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br><a href="http://www.cudatel.com">http://www.cudatel.com</a><br><br>Official FreeSWITCH Sites<br><a href="http://www.freeswitch.org">http://www.freeswitch.org</a><br><a href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a><br><a href="http://www.cluecon.com">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org">http://www.freeswitch.org</a><br><br><o:p></o:p></p><p class=MsoNormal><br><span style='font-size:11.0pt;font-family:"Courier New"'>-- <br>Ken<br><u><span style='color:blue'><a href="http://www.FreeSWITCH.org">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com">http://www.ClueCon.com</a><br><a href="http://www.OSTAG.org">http://www.OSTAG.org</a><br></span></u>irc.freenode.net #freeswitch</span><o:p></o:p></p></div></body></html>