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<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'>In Sip.conf I have the
following: </span></font><font size=2 face="Courier New"><span lang=EN-US
style='font-size:10.0pt;font-family:"Courier New"'>canreinvite=no <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Courier New"><span lang=EN-US
style='font-size:10.0pt;font-family:"Courier New"'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'>No, all telephones are on
the same subnet, handled by the same switch. I cannot verify if anything has
been changed since I installed & configured the network, but as far as I
know the whole network configuration is pretty straightforward, without any
routing madness.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'>Kind Regards,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'>Alex<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span lang=EN-US
style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=2 face=Tahoma><span lang=EN-US style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <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 4:34
PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-users] SIP
& Echo</span></font><span lang=EN-US><o:p></o:p></span></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>Are your
phones reinviting? Do you have any strange routing weirdness, or are they all
on a single subnet?<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
lang=EN-US style='font-size:12.0pt'>On 5/22/07, <b><span style='font-weight:
bold'>Asterisk</span></b> <<a href="mailto:asterisk@abraxas.si">
asterisk@abraxas.si</a>> wrote:</span></font></span><span lang=EN-US><o:p></o:p></span></p>
<div link=blue vlink=blue>
<div>
<p><font size=2 color=navy face=Arial><span lang=EN-US style='font-size:10.0pt;
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><span lang=EN-US><o:p></o:p></span></p>
<p><font size=2 color=navy face=Arial><span lang=EN-US style='font-size:10.0pt;
font-family:Arial;color:navy'> </span></font><span lang=EN-US><o:p></o:p></span></p>
<p><font size=2 color=navy face=Arial><span lang=EN-US style='font-size:10.0pt;
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><span lang=EN-US><o:p></o:p></span></p>
<p><font size=2 color=navy face=Arial><span lang=EN-US style='font-size:10.0pt;
font-family:Arial;color:navy'> </span></font><span lang=EN-US><o:p></o:p></span></p>
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<p><b><font size=2 face=Tahoma><span lang=EN-US style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span lang=EN-US style='font-size:10.0pt;font-family:Tahoma'> <a
href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">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> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-users] SIP
& Echo</span></font><span lang=EN-US><o:p></o:p></span></p>
</div>
<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'> <o:p></o:p></span></font></p>
<p style='margin-bottom:12.0pt'><font size=3 face="Times New Roman"><span
lang=EN-US style='font-size:12.0pt'>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. <o:p></o:p></span></font></p>
<div>
<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>On
5/22/07, <b><span style='font-weight:bold'>Asterisk</span></b> <<a
href="mailto:asterisk@abraxas.si" target="_blank">asterisk@abraxas.si</a>>
wrote:<o:p></o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>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">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">
asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Alexandre VERNIOL<br>
Sent: Tuesday, May 22, 2007 2:18 PM<br>
To: <st1:PersonName w:st="on">Asterisk Users Mailing List - Non-Commercial
Discussion</st1:PersonName><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>
> _______________________________________________<br>
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