<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">More info on my dropped call issue:<div><br></div><div>Here is a report on a dropped call from today:</div><div><br></div><div><div>Call Started echoing then cut out</div><div><br></div><div>Stats From the 7960 Stats Screen:</div><div><div>RxCnt:011853</div><div>TxCnt:010204</div><div>MaxJtr: 762</div><div>RxLost:0000</div></div><div><br></div><div><br></div><div>So, now I am starting to suspect that I have this problem:</div><div><br></div><div><a href="http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-ce-2-4-2-cisco-7960-7940-receiving-audio-drop-during-c">http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-ce-2-4-2-cisco-7960-7940-receiving-audio-drop-during-c</a></div><div><br></div><div>Any suggestions? </div><div><br></div><div>Mark</div><div><br></div></div><div><br></div><div><br><div><div>On Jul 10, 2009, at 2:45 PM, Steve Totaro wrote:</div><br><blockquote type="cite">DSS?<br><br>Ask them what other signaling they can support.<br><br>I would escalate every day if I were you. It is the only way to get things done. Get it to the top and be mad, even if you are not. <br><br>When fixed, PRAISE everyone from top to bottom.<br><br>A level one tech will say "Ah you are using Asterisk, we don't support that", or "Sorry, our switch cannot do 5ESS (when it can but takes a bit of work).<br><br>Thanks,<br>Steve Totaro<br><br><div> On Fri, Jul 10, 2009 at 2:24 PM, Mark Engelhardt <<a href="mailto:marke@intuitiveengineering.com">marke@intuitiveengineering.com</a>> wrote:<br> Conner,<br><br> I contacted my telco and they report they have a:<br><br> "EWSD Siemens Central Office"<br><br> Which does not support 5ess<br><br> Any other way around this? How did you determine changing to 5ess<br> would fix your problem?<br><br> Mark<br><div><div></div><div><br><br> On Jul 10, 2009, at 12:15 PM, Connor Spiess wrote:<br><br> > We had the same problem using a Digium T1 card. We switched the<br> > coding to from NI2 to 5ess and we haven't dropped a call since.<br> > You will have to check with your service provider to see if they do<br> > 5ess.<br> ><br> > Connor Spiess<br> > Network Specialist<br> ><br> ><br> > -----Original Message-----<br> > From: Mark Engelhardt [<a href="mailto:marke@intuitiveengineering.com">mailto:marke@intuitiveengineering.com</a>]<br> > Sent: Friday, July 10, 2009 10:58 AM<br> > To: Asterisk Users Mailing List - Non-Commercial Discussion<br> > Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas<br> > and a consultant.<br> ><br> > Hello Everyone.<br> ><br> > We have:<br> ><br> > Asterisk 1.4.21.2<br> > zaptel-1.4.11<br> > libpri-1.4.5<br> > Sangoma A101D Connected to a PRI<br> > Cicso 7960G phones (About 30 of them)<br> ><br> > We have a problem with dropped calls that has going on for a long<br> > time. We get up to 5 dropped calls on a bad day. They all seem to be<br> > incoming calls.<br> ><br> > I have a recording of what my users report a dropped call sounds like<br> > right before it drops<br> ><br> > <a href="http://www.stepawayfromthecomputer.com/drop.wav">http://www.stepawayfromthecomputer.com/drop.wav</a><br> ><br> > Please have a listen to the recording and tell me what you think it<br> > means<br> ><br> > I am looking for any ideas as to what I should do to track this down.<br> > I would love a lead on a good consultant who can help fix this.<br> ><br> > Mark<br> ><br> > _______________________________________________<br> > -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</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">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br> ><br> > ------------------------------------------------------------<br> > Mail was checked for spam by the Freeware Edition of No Spam Today!<br> > The Freeware Edition is free for personal and non-commercial use.<br> > You can remove this notice by purchasing a full license!<br> ><br> ><br> > _______________________________________________<br> > -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</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">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br><br> _______________________________________________<br> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</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">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></div></div></div><br><br clear="all"><br>-- <br>Thanks,<br>Steve Totaro <br>+18887771888 (Toll Free)<br>+12409381212 (Cell)<br>+12024369784 (Skype)<br> _______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</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">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div><br></div></body></html>