I have two links in my linkset which make it 60 channels.<br>what I am asking of is the whether the cause of my asterisk service stopping is due to bad oh323 or the version of asterisk i am using which is 1.2.5.<br><br>I have tested my chan_ss7 using h323callgen and I populated it with over 64 calls though without any voice and its was successfull.
<br><br>I am using oh323 version 0.7.3, asterisk 1.2.5 and latest version of chan_ss7.<br><br>goksie<br><br><div><span class="gmail_quote">On 4/28/06, <b class="gmail_sendername">Tom Chandler</b> <<a href="mailto:tchandle@eastex.net">
tchandle@eastex.net</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 style="direction: ltr;">
<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><div style="direction: ltr;"><span class="sg">
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2">Tom Chandler</font></div>
<div> </div>
</span></div><div style="direction: ltr;"><blockquote style="border-left: 2px solid rgb(0, 0, 0); padding-right: 0px; padding-left: 5px; margin-left: 5px; margin-right: 0px;"></blockquote></div><div style="direction: ltr;">
<span class="e" id="q_10ae1d19ea6042d6_3">
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;">----- Original Message ----- </div>
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<b>From:</b>
<a title="goksie@gmail.com" href="mailto:goksie@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Goke Aruna</a> </div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>To:</b> <a title="asterisk-ss7@lists.digium.com" href="mailto:asterisk-ss7@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
asterisk-ss7@lists.digium.com</a>
</div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>Sent:</b> Friday, April 28, 2006 1:14
PM</div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">anton.vazir@gmail.com</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;">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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