<div dir="ltr"><div>Ok so with some help from the IRC channel (thanks everyone!) I've managed to get pcaps of the calls and they sound fine and the max jitter is <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"><<a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@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 dir="ltr"><div><div>No matter what I try I can'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'm thinking of signing up with another SIP provider to test but I don't know of any other providers that offer pay as you go like Skype. Also as calls are fine I'm led to believe its not the Skype trunk at fault so don't want to waste money if I don'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"><<a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@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 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> "global_setvar zrtp_secure_media=false" 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"> <extension name="Local_Extension"></span><br style="font-family:arial,helvetica,sans-serif"><span style="font-family:arial,helvetica,sans-serif"> <condition field="destination_number" expression="^(10[01][0-9])$"></span><br style="font-family:arial,helvetica,sans-serif">
<span style="font-family:arial,helvetica,sans-serif"> <action application="export" data="dialed_extension=$1"/></span><br style="font-family:arial,helvetica,sans-serif"><span style="font-family:arial,helvetica,sans-serif"> <action application="playback" data="/home/user1/Downloads/8kulaw.wav"/></span><div>
<br style="font-family:arial,helvetica,sans-serif">
<span style="font-family:arial,helvetica,sans-serif"> <action application="voicemail" data="default $${domain} $1"/></span><br style="font-family:arial,helvetica,sans-serif"></div><span style="font-family:arial,helvetica,sans-serif"> </condition></span><br style="font-family:arial,helvetica,sans-serif">
<span style="font-family:arial,helvetica,sans-serif"> </extension></span><br><br></font></div><div><font>As I said before the wav plays fine then when routed to voiemail the audio issues occur.<br><br></font></div>
<div><font>Another paste <a href="http://pastebin.com/ZfWCzKYb" target="_blank">http://pastebin.com/ZfWCzKYb</a><br><br></font></div><div><font>Thanks!<br></font></div><div><font><br><br></font></div><div><br><br></div></div>
<div><div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Tue, Mar 19, 2013 at 3:40 PM, Yuriy Nasida <span dir="ltr"><<a href="mailto:nasida@live.ru" target="_blank">nasida@live.ru</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr"><div><font color="#444444"><span style="font-size:15px;line-height:21px">Did you add this line right after playing of 2 wav files you said ?</span></font></div><div><span style="color:rgb(68,68,68);font-size:15px;line-height:21px"> </span><font style="color:rgb(68,68,68);font-size:15px" face="arial,helvetica,sans-serif"><action application="voicemail" data="default $${domain} $1"/></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't tested from outside the LAN transitioning the external firewall, I'll give that a try.<br>
<br></div>I'm not sure about zrtp but I've changed the dialplan as you suggested to remove the loopback already. I'm using <span style="font-family:arial,helvetica,sans-serif"><action application="voicemail" data="default $${domain} $1"/></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"><<a href="mailto:nasida@live.ru" target="_blank">nasida@live.ru</a>></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 'loopback'.</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'm pretty new to FreeSwitch, how would I go about that? I'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"><<a href="mailto:nasida@live.ru" target="_blank">nasida@live.ru</a>></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'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>
<br><br><div>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 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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