<div dir="ltr"><div>Ok so with some help from the IRC channel (thanks everyone!) I&#39;ve managed to get pcaps of the calls and they sound fine and the max jitter is &lt;7ms.<br><br></div>Setting <span class=""><em>send_silence_when_idle <br>
</em></span></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Mar 20, 2013 at 6:02 PM, Elliot Farmer <span dir="ltr">&lt;<a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@gmail.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 dir="ltr"><div><div>No matter what I try I can&#39;t seem to get rid of these audio problems. Are there any known issues with Skype connect that anyone knows about?<br>
</div><br></div><div>I&#39;m thinking of signing up with another SIP provider to test but I don&#39;t know of any other providers that offer pay as you go like Skype. Also as calls are fine I&#39;m led to believe its not the Skype trunk at fault so don&#39;t want to waste money if I don&#39;t have to.<br>

<br></div><div>Is it worth installing on CentOS 5? The wiki lists some issues with 6 although I thought I would be safe on 6.3<br></div><div><br></div><div>Sorry to be asking so many questions, could just do with some pointers<br>

</div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Mar 20, 2013 at 11:36 AM, Elliot Farmer <span dir="ltr">&lt;<a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@gmail.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 dir="ltr"><div style="font-family:arial,helvetica,sans-serif"><font>Ok more testing and changes, still no closer.<br>

<br></font></div><div style="font-family:arial,helvetica,sans-serif"><font>I tested accessing the voicemail system externally through the firewall with a SIP phone and the IVR is perfectly clear, no problems.<br>
<br></font></div><div style="font-family:arial,helvetica,sans-serif"><font>I then disabled ZRTP by running this command</font><font> &quot;global_setvar zrtp_secure_media=false&quot; and from what I make of the log it worked however the issue remains.<br>


<br></font></div><div><font><span style="font-family:arial,helvetica,sans-serif">My dialplan entry looks like this...</span><br style="font-family:arial,helvetica,sans-serif"><br style="font-family:arial,helvetica,sans-serif">


<span style="font-family:arial,helvetica,sans-serif">    &lt;extension name=&quot;Local_Extension&quot;&gt;</span><br style="font-family:arial,helvetica,sans-serif"><span style="font-family:arial,helvetica,sans-serif">      &lt;condition field=&quot;destination_number&quot; expression=&quot;^(10[01][0-9])$&quot;&gt;</span><br style="font-family:arial,helvetica,sans-serif">


<span style="font-family:arial,helvetica,sans-serif">        &lt;action application=&quot;export&quot; data=&quot;dialed_extension=$1&quot;/&gt;</span><br style="font-family:arial,helvetica,sans-serif"><span style="font-family:arial,helvetica,sans-serif">        &lt;action application=&quot;playback&quot; data=&quot;/home/user1/Downloads/8kulaw.wav&quot;/&gt;</span><div>

<br style="font-family:arial,helvetica,sans-serif">
<span style="font-family:arial,helvetica,sans-serif">        &lt;action application=&quot;voicemail&quot; data=&quot;default $${domain} $1&quot;/&gt;</span><br style="font-family:arial,helvetica,sans-serif"></div><span style="font-family:arial,helvetica,sans-serif">      &lt;/condition&gt;</span><br style="font-family:arial,helvetica,sans-serif">


<span style="font-family:arial,helvetica,sans-serif">    &lt;/extension&gt;</span><br><br></font></div><div><font>As I said before the wav plays fine then when routed to voiemail the audio issues occur.<br><br></font></div>


<div><font>Another paste <a href="http://pastebin.com/ZfWCzKYb" target="_blank">http://pastebin.com/ZfWCzKYb</a><br><br></font></div><div><font>Thanks!<br></font></div><div><font><br><br></font></div><div><br><br></div></div>

<div><div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Tue, Mar 19, 2013 at 3:40 PM, Yuriy Nasida <span dir="ltr">&lt;<a href="mailto:nasida@live.ru" target="_blank">nasida@live.ru</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">





<div><div dir="ltr"><div><font color="#444444"><span style="font-size:15px;line-height:21px">Did you add this line right after playing of 2 wav files you said ?</span></font></div><div><span style="color:rgb(68,68,68);font-size:15px;line-height:21px"> </span><font style="color:rgb(68,68,68);font-size:15px" face="arial,helvetica,sans-serif">&lt;action application=&quot;voicemail&quot; data=&quot;default $${domain} $1&quot;/&gt;</font><div>


<font color="#444444" face="arial, helvetica, sans-serif"><span style="font-size:15px"><br></span></font></div></div><div><font color="#444444" face="arial, helvetica, sans-serif"><span style="font-size:15px">Also you can check </span></font></div>


<div><a href="http://wiki.freeswitch.org/wiki/ZRTP#Step_2" target="_blank">http://wiki.freeswitch.org/wiki/ZRTP#Step_2</a><font color="#444444" face="arial, helvetica, sans-serif"><span style="font-size:15px"><br></span></font><br>


<div><div></div><hr>Date: Tue, 19 Mar 2013 15:18:12 +0000<div><div><br>From: <a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@gmail.com</a><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>


Subject: Re: [Freeswitch-users] External voicemail issue<br><br><div dir="ltr"><div>From a SIP handset directly to FreeSwitch voicemail works perfectly, although I haven&#39;t tested from outside the LAN transitioning the external firewall, I&#39;ll give that a try.<br>


<br></div>I&#39;m not sure about zrtp but I&#39;ve changed the dialplan as you suggested to remove the loopback already. I&#39;m using <span style="font-family:arial,helvetica,sans-serif">&lt;action application=&quot;voicemail&quot; data=&quot;default $${domain} $1&quot;/&gt;</span>
instead of the loopback but the issue is still there.<br><pre><br><br></pre></div><div><br><br><div>On Tue, Mar 19, 2013 at 3:07 PM, Yuriy Nasida <span dir="ltr">&lt;<a href="mailto:nasida@live.ru" target="_blank">nasida@live.ru</a>&gt;</span> wrote:<br>



<blockquote style="border-left:1px #ccc solid;padding-left:1ex">


<div><div dir="ltr"><div>Also looks like that you use  zrtp.</div><div><br></div><div>Um.. Skype + loopback + zrtp + TRANSCODING.</div><div><span style="font-size:12pt">I can just advise you to simplify your setup and look that is the cause.</span></div>



<div>Can you try with SIP-to-SIP directly to voicemail and without any additional things  ?</div><br><div><div></div><hr>From: <a href="mailto:nasida@live.ru" target="_blank">nasida@live.ru</a><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>



Date: Tue, 19 Mar 2013 18:24:51 +0400<div><div><br>Subject: Re: [Freeswitch-users] External voicemail issue<br><br>


<div dir="ltr"><div><a href="http://wiki.freeswitch.org/wiki/Loopback" target="_blank">http://wiki.freeswitch.org/wiki/Loopback</a></div><div><br></div><div>You can look at your dial plan and look for &#39;loopback&#39;.</div>



<div><span style="font-size:12pt">the loopback endpoint affects to voice and if you really use this it can be the reason of your issue.</span></div><div><span style="font-size:12pt">In this case in my opinion a redoing of your routing logic without the loopback endpoint will be good idea.</span></div>



<div><br></div><div><br><div><div></div><hr>Date: Tue, 19 Mar 2013 13:55:34 +0000<br>From: <a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@gmail.com</a><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>



Subject: Re: [Freeswitch-users] External voicemail issue<br><br><div dir="ltr"><div>Sorry I&#39;m pretty new to FreeSwitch, how would I go about that? I&#39;m using the default config that has loopback setup as the route to voicemail as far as I can work out.<br>



<br></div>Is there a way of sending to voicemail without a loopback?<br>
<div><div><div><br><br><div>On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida <span dir="ltr">&lt;<a href="mailto:nasida@live.ru" target="_blank">nasida@live.ru</a>&gt;</span> wrote:<br>
<blockquote style="border-left:1px #ccc solid;padding-left:1ex">


<div><div dir="ltr"><div><font face="sans-serif"><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></div><hr>Date: Tue, 19 Mar 2013 12:37:47 +0000<br>From: <a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@gmail.com</a><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>




Subject: Re: [Freeswitch-users] External voicemail issue<div><div><br><br><div dir="ltr"><div><div><div>I&#39;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&#39;ve tried preferring the PCMU codec within vars but I seem to have a lot of 
&quot;TRANSCODING_NECESSARY&quot; 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>
<br><br><div>On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer <span dir="ltr">&lt;<a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@gmail.com</a>&gt;</span> wrote:<br><blockquote 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&#39;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 &quot;the recording is below the minimum length&quot; 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&#39;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&#39;ve tried to remove the phone numbers and IP addresses as I don&#39;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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<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div></div></div></div>                                               </div></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
<a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
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