Use <a href="http://fisheye.freeswitch.org">fisheye.freeswitch.org</a> or do a git checkout and &quot;git log -p&quot;<br>-MC<br><br><div class="gmail_quote">On Wed, Dec 19, 2012 at 4:14 PM, Yungwei Chen <span dir="ltr">&lt;<a href="mailto:yungwei@resolvity.com" target="_blank">yungwei@resolvity.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div link="blue" vlink="purple" lang="EN-US"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Please tell me the list of files affected by this fix or the bug id. Thanks.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u> <u></u></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:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">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<u></u><u></u></span></p></div><p class="MsoNormal"><u></u> <u></u></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<u></u><u></u></p><div><p class="MsoNormal">On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen &lt;<a href="mailto:yungwei@resolvity.com" target="_blank">yungwei@resolvity.com</a>&gt; wrote:<u></u><u></u></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&#39;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>    &lt;extension name=&quot;public_extensions&quot;&gt;<br>      &lt;condition field=&quot;destination_number&quot; expression=&quot;^\d{10}$&quot;&gt;<br>
        &lt;action application=&quot;transfer&quot; data=&quot;main XML default&quot;/&gt;<br>      &lt;/condition&gt;<br>    &lt;/extension&gt;<br><br>2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context.<br>
  &lt;extension name=&quot;test&quot;&gt;<br>      &lt;condition field=&quot;${destination_number}&quot; expression=&quot;^main$&quot;&gt;<br>        &lt;action application=&quot;set&quot; data=&quot;RECORD_STEREO=false&quot;/&gt;<br>
        &lt;action application=&quot;answer&quot; /&gt;<br>        &lt;action application=&quot;record_session&quot; data=&quot;/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav&quot;/&gt;<br>
        &lt;action application=&quot;bridge&quot; data=&quot;{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890&quot;/&gt;<br>      &lt;/condition&gt;<br>  &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>
    &lt;include&gt;<br>        &lt;gateway name=&quot;gw1&quot;&gt;<br>          &lt;param name=&quot;username&quot; value=&quot;&quot;/&gt;<br>          &lt;param name=&quot;password&quot; value=&quot;&quot;/&gt;<br>          &lt;param name=&quot;realm&quot; value=&quot;&quot;/&gt;<br>
          &lt;param name=&quot;from-domain&quot; value=&quot;&quot;/&gt;<br>          &lt;param name=&quot;extension&quot; value=&quot;&quot;/&gt;<br>          &lt;param name=&quot;expire-seconds&quot; value=&quot;60&quot;/&gt;<br>
          &lt;param name=&quot;register&quot; value=&quot;false&quot;/&gt;<br>          &lt;param name=&quot;retry-seconds&quot; value=&quot;60&quot;/&gt;<br>        &lt;/gateway&gt;<br>    &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>
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</div><span class="HOEnZb"><font color="#888888"><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>
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<br></blockquote></div><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>
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