Latest version of FreeSWITCH has some updates that may fix this issue. I would update to 1.2.5.3 ASAP.<br>-MC<br><br><div class="gmail_quote">On Tue, Dec 18, 2012 at 1:56 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">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>
<br>
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</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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