Are your phones reinviting? Do you have any strange routing weirdness, or are they all on a single subnet?<br><br><div><span class="gmail_quote">On 5/22/07, <b class="gmail_sendername">Asterisk</b> <<a href="mailto:asterisk@abraxas.si">
asterisk@abraxas.si</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div link="blue" vlink="blue" lang="SL">
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<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">I tried with the ping ... all of the
phones respond in ca. 0.3ms, so network seems to be OK. More than 90% of CPU on
* box is idle even in peak times, so this shouldn't cause echoes either, right?
Hmmm, so handset could be an issue, but did anyone ever experience any handset
problems with Polycom IP SoundPoint 430 :-) ?</span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">I will check the headsets and any
possibilities of acoustical echo. Besides that, if we rule out the network, and
the CPU on the * box, is there anything else that could be causing echoes on
internal SIP calls?</span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
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<p><b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;" lang="EN-US">From:</span></font></b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>] <b><span style="font-weight: bold;">On Behalf Of
</span></b>David Gomillion<br>
<b><span style="font-weight: bold;">Sent:</span></b> Tuesday, May 22, 2007 3:22
PM<br>
<b><span style="font-weight: bold;">To:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion<br>
<b><span style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users] SIP
& Echo</span></font><span lang="EN-US"></span></p>
</div>
<p><font face="Times New Roman" size="3"><span style="font-size: 12pt;"> </span></font></p>
<p style="margin-bottom: 12pt;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">We experience echo too
from time to time. It's usually headset-related, but not always. I ran a
persistent ping on one of the phones, and we diagnosed a wiring problem with
it. Other phones needed a new handset. The problem is that these problems need
to be fixed on the phone NOT hearing echo. </span></font></p>
<div>
<p><span><font face="Times New Roman" size="3"><span style="font-size: 12pt;">On 5/22/07, <b><span style="font-weight: bold;">Asterisk</span></b>
<<a href="mailto:asterisk@abraxas.si" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk@abraxas.si</a>> wrote:</span></font></span></p>
<p><font face="Times New Roman" size="3"><span style="font-size: 12pt;">How could I check if bandwith or/and latency is causing it?<br>
<br>
If I do SIP show peers it says OK (13 ms) for all peers. I guess there is a way
to gather more detailed info on SIP calls and latency?<br>
<br>
* box is connected to the 1Gb switch with 1Gb connection, and clients have
100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP hardphones
connected to the * box. <br>
<br>
Thanks, Alex<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Alexandre VERNIOL<br>
Sent: Tuesday, May 22, 2007 2:18 PM<br>
To: Asterisk Users Mailing List - Non-Commercial
Discussion<br>
Subject: Re: [asterisk-users] SIP & Echo <br>
<br>
Hi,<br>
<br>
Could be bandwith or/and latency ... Many causes...<br>
<br>
<br>
Alex<br>
<br>
Asterisk a écrit :<br>
> Hello all,<br>
><br>
> One of our clients reported that they are experiencing echo in SIP calls<br>
> (even on internal ones). What do you think could be causing echo in<br>
> internal SIP calls?<br>
><br>
> We're using Polycom telephones, do you think they could be causing it?<br>
><br>
> Thanks,<br>
> Alex <br>
><br>
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