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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I’ve checked the source code of
chan_ss7.c and it doesn’t have any calls to turn the echo canceller on,
is any workaround available in the meantime the next release of chan_ss7 is
released?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span 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 style='font-size:
10.0pt;font-family:Arial;color:navy'>Greets, <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span 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 style='font-size:
10.0pt;font-family:Arial;color:navy'>Luciano<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span 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 style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-ss7-bounces@lists.digium.com
[mailto:asterisk-ss7-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Tom Chandler<br>
<b><span style='font-weight:bold'>Sent:</span></b> Viernes, 28 de Abril de 2006
03:42 p.m.<br>
<b><span style='font-weight:bold'>To:</span></b> asterisk-ss7@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-ss7]
asterisk oh323 - chan-ss7 echo problem</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>If you are using SS7 and E1, then one channel is taken for
signaling, leaving 30 channels for voice. SS7 will not send more calls
than available channels. You should be getting a message on the console
stating no CIC was available........</span></font><o:p></o:p></p>
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<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Tom Chandler</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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margin-left:3.15pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>----- Original Message ----- <o:p></o:p></span></font></p>
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<p class=MsoNormal style='background:#E4E4E4'><b><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial;font-weight:bold'>From:</span></font></b><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'> <a
href="mailto:goksie@gmail.com" title="goksie@gmail.com">Goke Aruna</a> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><b><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial;font-weight:bold'>To:</span></font></b><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'> <a
href="mailto:asterisk-ss7@lists.digium.com"
title="asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><b><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial;font-weight:bold'>Sent:</span></font></b><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'> Friday, April 28,
2006 1:14 PM<o:p></o:p></span></font></p>
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<div>
<p class=MsoNormal><b><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial;font-weight:bold'>Subject:</span></font></b><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'> Re: [asterisk-ss7]
asterisk oh323 - chan-ss7 echo problem<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Can someone suggest any
other h323 protocol that could<br>
use with asterisk instead of oh323?<br>
<br>
I had similar problem with oh323 and each time it happens the asterisk will
stop at a call going above 30 simultaneus calls <br>
<br>
However, I read from voip-info and which give oh323 some ratings to oh323.<br>
<br>
I used A104D sangoma card with chan_ss7 and no echo.<br>
<br>
goksie<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 4/28/06, <b><span style='font-weight:bold'>Anton</span></b>
<<a href="mailto:anton.vazir@gmail.com">anton.vazir@gmail.com</a>> wrote:</span></font></span>
<o:p></o:p></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Dear Jacob,<br>
<br>
Any chance to give us any timing approximation for a next<br>
version?<br>
<br>
Regards,<br>
Anton.<br>
<br>
On 28 April 2006 17:26, Jacob Tinning wrote:<br>
> On Tue, 18 Apr 2006, leonimar cape wrote:<br>
> > Can someone give a suggestion on what I should do to <br>
> > omit the echo. Here is may scenario<br>
> ><br>
> > --- h323
--- chan-ss7 ---<br>
>
> A---| |--------| |--------------| |----B<br>
>
> ---
---
--- <br>
> >
Nextone Asterisk
DMS<br>
><br>
> The next version of chan_ss7 will include<br>
> enabling/disabling of the zaptel echo-canceller, which<br>
> probably will solve your problem.<br>
><br>
> > The calling party can hear every words he/she say<br>
> > after 1 to 2 seconds. But the called party can hear A<br>
> > with no echo and the quality is clear. I have already<br>
> > tried it in both digium (TE410P) and sangoma card <br>
> > (AT104) and the results where the same. Is there any<br>
> > way that I can cancel the echo? Any particular<br>
> > settings that I have to change in the settings of the<br>
> > asterisk?<br>
> <br>
> The current version of chan_ss7 does not do anything to<br>
> avoid echo. It just passes the audio through the<br>
> channels.<br>
><br>
> The problem is when A calls an old analog phone B. Old<br>
> analog phones typically bleed some of the audio back to <br>
> the sender.<br>
><br>
> Ordinary synchronous telephone-networks doesn't delay the<br>
> audio very much ( < 10ms) so the caller will not notice<br>
> any echo.<br>
><br>
> Unfortunately, ip-networks induce more delay ( > 70 ms ) <br>
> which will clearly sound as echo.<br>
><br>
> To avoid the echo, the only (as far as I know) solution<br>
> is to insert some kind of echo-canceller "in" or "to the<br>
> right" of Asterisk in your drawing. <br>
><br>
> As noted above, the next version of chan_ss7 will start<br>
> the zaptel echo-cancellation when a new call is made<br>
> (this is configurable, if you don't want<br>
> echo-cancellation), and stop it again when the call is <br>
> finished.<br>
><br>
> Mvh. Jacob<br>
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