In this connection there are min of 2 hybrid circuits on my end<br>
<br>
Vonage ATA box ---- SIPURA ATA ---- Asterisk ----- SIPURA ATA (phone)<br>
hybird 1 ----- 2 wire -- hybird 2 ----------------------hybird 3 (if not 841 phone)<br>
<br>
I also think vonage has one more hybird on there end.<br>
<br>
Since the connection to vonage is 2 ata back to back this means 2 hybirds. The ata are<br>
connect thought 2 wire interface.<br><br><div><span class="gmail_quote">On 12/18/05, <b class="gmail_sendername">Mohammad Shokuie</b> <<a href="mailto:shokuie@hotmail.com">shokuie@hotmail.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 pals,<br><br>As a matter of fact im serious to know where is the source of echo in a pure<br>VoIP connection, i think the most of echo problems come from hybrid circuits<br>which are not an issue in pure VoIP sessions.
<br><br>Regards.<br>---<br>M. Shokuie Nia.<br><br><br>>From: Luki <<a href="mailto:lugosoft@gmail.com">lugosoft@gmail.com</a>><br>>Reply-To: Asterisk Users Mailing List - Non-Commercial<br>>Discussion<<a href="mailto:asterisk-users@lists.digium.com">
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asterisk-users-bounces@lists.digium.com</a><br>>X-OriginalArrivalTime: 18 Dec 2005 05:46:37.0982 (UTC)<br>>FILETIME=[69B0F3E0:01C60396]<br>><br>> > Before I start hacking this into asterisk 1.2.1 I would like to known
<br>> > if others are running into this kind of problem ?<br>><br>>Asterisk doesn't do any echo cancellation in the setup you describe;<br>>it just passes the audio data, and transcodes if necessary. The<br>
>endpoints (the 841 phone and the 2002 and 3000 ATAs) are responsible<br>>for cancelling echo.<br>><br>>The Sipura ATA's generally do a good job cancelling echo. You may want<br>>to play with the gain settings in the admin web config for the Sipura
<br>>ATA. As far as the 841 is concerned, if the handset volume is too loud<br>>I noticed you may be getting acoustic echo. Hasn't been a problem for<br>>me for PSTN calls or SIP to SIP calls though.<br>><br>>If you really want to patch asterisk to apply echo cancellation on the
<br>>RTP stream on pure VoIP calls, that would be interesting to see how<br>>well it works.<br>><br>>--Luki<br>>_______________________________________________<br>>--Bandwidth and Colocation provided by
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