Well the phones I am using are Linksys SPA942. The Sip trunking I am using is Flowroute.<div><br></div><div>The problem only happens with calls from flowroute to me. Then both the flowroute side and the SPA942 side are fast on the recording.</div>
<div><br></div><div>START</div><div>Flowroute ------> Freeswitch -----> SPA942</div><div><br></div><div>If the call goes from SPA942 to flowroute then it sounds just fine in the recording.</div><div><br></div><div>START</div>
<div>SPA942 ----> Freeswitch ------> Flowroute.</div><div><br></div><div>I have done a git and I am on the most current release </div><div><br></div><div><div>FreeSWITCH Version 1.3.0+git~20120908T211235Z~36cee285b0 (1.3.0; git at commit 36cee285b0 on Sat, 08 Sep 2012 21:12:35 Z)</div>
</div><div><br><div><br><div class="gmail_quote">On Sun, Sep 9, 2012 at 12:54 AM, Vbvbrj <span dir="ltr"><<a href="mailto:vbvbrj@gmail.com" target="_blank">vbvbrj@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="im">On 09.09.2012 05:40, Joshua Foshee wrote:<br>
> We have a issue with the call quality of the recorded call to be in a<br>
> faster speed then what the call takes place in. I have took a packet<br>
> capture on the server to replay the audio stream and it sounds just<br>
> fine. Anyone have the same problem with the speed of the audio recording<br>
<br>
</div>This happens because of transcoding. If you look carefully, the bad<br>
audio will be only for certain users' sip phones. Look for phones<br>
connected to problematic extensions and change in phone settings the<br>
signaling standard from china to other. Also update FS to latest 1.3.0<br>
release.<br>
<br>
--<br>
Mimiko desu.<br>
<br>
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</blockquote></div><br></div></div>