<p>Asterisk is a swiss army knife, you should either know how to use it or rely on ready made software which control routing of calls through variable bit rates (skype does that very effectively)</p>
<p>So the key here for you to research upon from those several hundred results is "variable bit rate codec negotiations"</p>
<p>Mitul Limbani<br>
<a href="http://www.facebook.com/enterux">www.facebook.com/enterux</a><br>
<a href="http://www.facebook.com/entvoice">www.facebook.com/entvoice</a><br>
</p>
<div class="gmail_quote">On Oct 29, 2013 1:30 AM, "Ron Wheeler" <<a href="mailto:rwheeler@artifact-software.com">rwheeler@artifact-software.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>I am reaching the same level of
frustration.<br>
I have tried to find the source of the problems. <br>
We have IAX2 to our VoIP provider and SIP phones attached to the
Asterisk - No analogue.<br>
We have a very lightly loaded 60 Mbs cable link to the Internet
that tests pretty close to that most of the time.<br>
<br>
I have not found any good tools to track down the causes of poor
voice quality.<br>
In my case, I have good incoming quality and terrible quality
going out.<br>
That is, I can hear people perfectly well but they complain that
my voice drops out and is garbled regardless of who places the
call.<br>
As a result, I use Skype for all of my calls and if someone calls
me, I call them back on Skype if they have any problems.<br>
I don't understand why Skype works so well and Asterisk works so
poorly on the same environment.<br>
<br>
Googling "Asterisk poor audio quality" return several hundred
thousand references <br>
<br>
Ron<br>
On 28/10/2013 2:29 PM, Eddie Mikell wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">All,
<div><br>
</div>
<div>The users in our organization are well, quite frankly, sick
of phone service that is being provided. The choppy phone
calls, and drop outs are detrimental to our sales force.</div>
<div>
<br>
</div>
<div>I've tried about everything I can think of. </div>
<div><br>
</div>
<div>
<blockquote style="margin:0 0 0 40px;border:none;padding:0px">
<div>Moved the asterisk server from VM machine to dedicated
machine</div>
</blockquote>
<blockquote style="margin:0 0 0 40px;border:none;padding:0px">
<div>More than enough bandwidth</div>
</blockquote>
<blockquote style="margin:0 0 0 40px;border:none;padding:0px">
<div>Setting 802.1p = 7</div>
</blockquote>
<blockquote style="margin:0 0 0 40px;border:none;padding:0px">
<div>Set Dedicated voice traffic 35% of bandwidth.</div>
<div><br>
</div>
</blockquote>
Not sure what option would be the best</div>
<div>
<blockquote style="margin:0 0 0 40px;border:none;padding:0px">
<div><br>
</div>
</blockquote>
<blockquote style="margin:0 0 0 40px;border:none;padding:0px">
<div>Put analog lines in the conference room to avoid the
dropouts - leave the sip lines in place for day to day use</div>
</blockquote>
<blockquote style="margin:0 0 0 40px;border:none;padding:0px">
<div>Hire a consultant</div>
</blockquote>
<blockquote style="margin:0 0 0 40px;border:none;padding:0px">
<div>Ditch the system and buy a pre-packaged system -
RingCentral or some such.</div>
<div><br>
</div>
</blockquote>
<div>There are no local asterisk professionals who can help,
and we are a little leery of opening up our system to
outside consultants.</div>
<div><br>
</div>
Anyone else face the above, and finally abandoned Asterisk for
a commercial system? </div>
<div><br>
</div>
<div>We have 167 users.<br>
<div>I use Grandstream GXP 2100 on the desktop and Polycom
ip6000 for the conference rooms.</div>
<div><br>
</div>
<div>Suggestions welcome.</div>
<div><br>
</div>
<div>Best</div>
<div>
<br>
</div>
<div>Eddie</div>
-- <br>
<div dir="ltr">Eddie H. Mikell
<div>Senior Systems Engineer</div>
<div>RKG</div>
<div><br>
</div>
<div>Office: <a href="tel:434.970.1010%20x%20124" value="+914349701010" target="_blank">434.970.1010 x 124</a></div>
<div>Email:<a href="mailto:emikell@rimmkaufman.com" target="_blank">
emikell@rimmkaufman.com</a></div>
</div>
</div>
</div>
<br>
<a href="http://www.rimmkaufman.com" target="_blank"><img src="http://www.rimmkaufman.com/content/RKG_web1380.gif"></a>
<div><a href="http://twitter.com/rimmkaufman" target="_blank"><img src="http://resources.rimmkaufman.com/rs/rimmkaufmangroup/images/twit-icon-a.gif"></a> <a href="http://www.linkedin.com/company/85385" target="_blank"><img src="http://resources.rimmkaufman.com/rs/rimmkaufmangroup/images/linked-icon-a1.gif"></a> <a href="http://plus.google.com/104980442218952272663/posts" target="_blank"><img src="http://resources.rimmkaufman.com/rs/rimmkaufmangroup/images/goog%2B-icon-a.gif"></a> <a href="http://www.facebook.com/rimmkaufman" target="_blank"><img src="http://resources.rimmkaufman.com/rs/rimmkaufmangroup/images/fb-icon-a.gif"></a> <a href="http://www.RKGblog.com" target="_blank"><img src="http://www.rimmkaufman.com/content/rkgblog18x75.png"></a><br>
<p style="font-size:12.800000190734863px;font-family:arial,sans-serif;margin:0px"><br>
</p>
</div>
<br>
<fieldset></fieldset>
<br>
</blockquote>
<br>
<br>
<pre cols="72">--
Ron Wheeler
President
Artifact Software Inc
email: <a href="mailto:rwheeler@artifact-software.com" target="_blank">rwheeler@artifact-software.com</a>
skype: ronaldmwheeler
phone: 866-970-2435, ext 102</pre>
</div>
<br>--<br>
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