<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:'times new roman', 'new york', times, serif;font-size:12pt"><div>10%</div><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><br><div style="font-family:arial, helvetica, sans-serif;font-size:13px"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight: bold;">De:</span></b> Matt Riddell <lists@venturevoip.com><br><b><span style="font-weight: bold;">Para:</span></b> asterisk-users@lists.digium.com<br><b><span style="font-weight: bold;">Enviadas:</span></b> Quarta-feira, 4 de Maio de 2011 0:32:28<br><b><span style="font-weight: bold;">Assunto:</span></b> Re: [asterisk-users] Fading voice problem<br></font><br>On 3/05/11 10:16 PM, Eduardo Leones wrote:<br>> Guys,<br>><br>> I'm having problems in the fading voice calls, receptive and active,<br>> that in SIP accounts. While few people using
the system, calls are<br>> perfect, but it beats the normal use of connections (average 30<br>> concurrent), the voice begins to fade from people.<br>><br>> Soon I figured some network problem, I did a tcpdump and analyzed by<br>> wireshark ...the strange thing is this ...<br>><br>> all packets that arrive on the server asterisk are normal or jitter,<br>> latency ... But whenAsterisk sends packets to the network or the ISP ...<br>> maggoty packages are ... jitter of150ms on average ... latency of more<br>> than 1000 ms ...<br>><br>> That is, by the way is not the network itself, but the network on the<br>> machine ...<br>><br>> Dropped iptables to make sure no influence ... I changed the network<br>> card and cables... did nothing more ...<br>><br>> Anyone have any ideas to help me and chase to find the problem?<br>><br>> PS: The server is a CentOS 5.5 - 32 bit ... I've tested the 64bit tb
but<br>> with the sameerror ...<br><br>What's your CPU usage like?<br><br>-- <br>Cheers,<br><br>Matt Riddell<br>_______________________________________________<br><br><a href="http://www.venturevoip.com/news.php" target="_blank">http://www.venturevoip.com/news.php</a> (Daily Asterisk News)<br><a href="http://www.venturevoip.com/exchange.php" target="_blank">http://www.venturevoip.com/exchange.php</a> (Full ITSP Solution)<br><a href="http://www.venturevoip.com/cc.php" target="_blank">http://www.venturevoip.com/cc.php</a> (Call Centre Solutions)<br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello"
target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></div></div><div style="position:fixed"></div>
</div></body></html>