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<div class="moz-cite-prefix">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
cite="mid:CAO12vcvwHJbFBpa0Zsi3sb9gK7vUJt4_4x3qmdDNjufnYxd5Yw@mail.gmail.com"
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: 434.970.1010 x 124</div>
<div>Email:<a moz-do-not-send="true"
href="mailto:emikell@rimmkaufman.com" target="_blank">
emikell@rimmkaufman.com</a></div>
</div>
</div>
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<pre class="moz-signature" cols="72">--
Ron Wheeler
President
Artifact Software Inc
email: <a class="moz-txt-link-abbreviated" href="mailto:rwheeler@artifact-software.com">rwheeler@artifact-software.com</a>
skype: ronaldmwheeler
phone: 866-970-2435, ext 102</pre>
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