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<body class='hmmessage'><div dir='ltr'><div><font face="sans-serif" size="2"><span style="line-height: 19.046875px;">looked at your logs. Do you use loopback endpoint in your dialplan ? If yes, can you check without this ?</span></font></div><br><div><div id="SkyDrivePlaceholder"></div><hr id="stopSpelling">Date: Tue, 19 Mar 2013 12:37:47 +0000<br>From: elliotfarmer101@gmail.com<br>To: freeswitch-users@lists.freeswitch.org<br>Subject: Re: [Freeswitch-users] External voicemail issue<br><br><div dir="ltr"><div><div><div>I've done some further testing and the issue appears to be isolated to the voicemail system.<br><br></div>To test I configured the inbound dialplan to play two wav files before transferring the call...<br>
<br>1) 8kz wav from wikipedia <a href="http://www.nch.com.au/acm/8kulaw.wav" target="_blank">http://www.nch.com.au/acm/8kulaw.wav</a><br>2) /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav<br><br></div>Both play fine, as soon as the call is transferred to voicemail (via extension 1000 that is not registered at the time) the quality drops and I start loosing speech. I've tried preferring the PCMU codec within vars but I seem to have a lot of
"TRANSCODING_NECESSARY" messages, they are not isolated to the
voicemail steps however.<br><br></div>Has anyone got an idea where I can start to look? Any help would be much appreciated.<br><div><div><div><div><div><div><div><div><div><div><div><div><div><div><div class="ecxgmail_extra">
<br><br><div class="ecxgmail_quote">On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer <span dir="ltr"><<a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@gmail.com</a>></span> wrote:<br><blockquote class="ecxgmail_quote" style="border-left:1px solid rgb(204,204,204);padding-left:1ex;">
<div dir="ltr"><div><div><div><div><div><div>Hi all,<br><br></div>I'm having an issue with the voicemail system when accessing via a Skype connect trunk. <br><br></div>Internally all is working, however if I call my external Skype number from a PSTN line when the call is routed to voicemail I get a lot of intermittent speech loss during the IVR announcement and then after the beep I can hear odd sounds that I can only describe as digital interference type sounds. Also the IVR is very quick to say "the recording is below the minimum length" although if I speak quickly I can record a voicemail and it sounds fine during playback, not amazing audio quality but no major issues.<br>
<br></div>The issue seems to be the same as <a href="http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html" target="_blank">http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html</a>, I have tried recording a greeting but the same issues occur.<br>
<br></div>I'm running Freeswitch on CentOS 6.3 on a physical machine.<br><br></div>Here are some logs <a href="http://pastebin.com/jmuks47q" target="_blank">http://pastebin.com/jmuks47q</a> , I've tried to remove the phone numbers and IP addresses as I don't have permission from the owners to publish them on the internet.<br>
<br></div>Thanks in advance for your help! <br><div><div><div><br><br></div></div></div></div>
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