<div dir="ltr">Changing clock to soft is causing call latency issues. Is there any other way? It works well with Asterisk with no timer.</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Dec 17, 2013 at 2:39 PM, Peter Olsson <span dir="ltr"><<a href="mailto:peter@olssononline.se" target="_blank">peter@olssononline.se</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">Well, playback without a timer won't work for you, so stick to the soft timer.</div><div class="HOEnZb">
<div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">2013/12/17 Ashwin Jain <span dir="ltr"><<a href="mailto:ashwinrkjain@gmail.com" target="_blank">ashwinrkjain@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">It is stored locally. <div>It works fine with "soft" timer (using this we had call latency issues), but have this issue when I use "none" as rtp-timer-name.</div>
<div class="gmail_extra">
<br><br><div class="gmail_quote">On Tue, Dec 17, 2013 at 10:00 AM, Ali Pey <span dir="ltr"><<a href="mailto:alipey@gmail.com" target="_blank">alipey@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">Is the welcome message stored locally or is it somewhere on NAS?<div><br></div><div>Regards,</div><div>
Ali Pey<br>
<div class="gmail_extra"><br><br><div class="gmail_quote"><div><div>On Sun, Dec 15, 2013 at 1:42 AM, Ashwin Jain <span dir="ltr"><<a href="mailto:ashwinrkjain@gmail.com" target="_blank">ashwinrkjain@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">Hi,<div><br></div><div>We are facing issues with the bad audio quality (audio is sounding like its recorded underwater) only for the <b>playback application</b>. There is no issue with the audio call quality (after bridging). </div>
<div><br></div><div>Audio file used with playback application is recorded at 8khz with 16 bits/sample and 1 channel. <br clear="all"><div><br></div><div>Following is the pseudo dialplan:</div><div>1. Call comes on freeswitch server.</div>
<div>2. Call answered.</div><div>3. Query database to identify destination</div><div>4. Play a welcome message audio (using playback application)</div><div>5. Start call recording</div><div>6. Bridge the call</div><div><br>
</div><div>In step 4, we are facing this issue.</div><span><font color="#888888">-- <br><div dir="ltr">Thanks and Regards,<br>Ashwin Jain<div>MetroGuild</div></div>
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<br></blockquote></div><span><font color="#888888"><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Thanks and Regards,<br>Ashwin Jain<br>Director (Engineering)<div>MetroGuild</div></div>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Thanks and Regards,<br>Ashwin Jain<br>Director (Engineering)<div>MetroGuild</div></div>
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