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<DIV><FONT face=Arial size=2>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........</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Tom Chandler</FONT></DIV>
<DIV> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=goksie@gmail.com href="mailto:goksie@gmail.com">Goke Aruna</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-ss7@lists.digium.com
href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, April 28, 2006 1:14
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [asterisk-ss7] asterisk
oh323 - chan-ss7 echo problem</DIV>
<DIV><BR></DIV>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<BR><BR>
<DIV><SPAN class=gmail_quote>On 4/28/06, <B class=gmail_sendername>Anton</B>
<<A href="mailto:anton.vazir@gmail.com">anton.vazir@gmail.com</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">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>_______________________________________________<BR>--Bandwidth and
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