Below is the text of my original post. I am not sure what Codec we are using. The "Codec Preferences" phone setting shows, in order of preference, G.711u, G.711A, G.729AB<br> <br> <div class="MsoNormal">We are running asterisk-1.2.4 with zaptel-1.2.7 on Fedora Core 4-2.6.14-1.1656_FC4smp.<span style=""> </span>It is<span style=""> </span>installed on a Dell PE 2500 with 2x900 MHz processors and 1 Gb RAM and 1 SCSI Disk. The server has a Digium TDM400P card which is connected to 4 POTS lines.<span style=""> </span>The server is also connected to a 100MB switched LAN where we have about 20 Polycom 501 phones with the latest firmware updates. Nothing else runs on the server except an ftp daemon which is never used except when a phone reboots.<br> <br> For about 20% of the calls to the outside world, the voice on the other end of an outside line is incredibly choppy.<span style=""> </span>Enough to where we have to hang up and
call on a cell phone. It is always the same numbers that are choppy.<span style=""> </span>The funny thing is, if I press mute while talking on a choppy call, the choppiness goes away completely.</div> <div class="MsoNormal"> I have tried: turning off ACPI, turning off APCI, moving the card to another PCI slot, changing the RX/TX gains. There are no shared IRQs. I have tested the lines by unplugging them from the asterisk server and plugging them directly into an analogue phone. Using "cat /proc/interrupts; sleep 10 ; cat /proc/interrupts" I see that there are about 1,000 interrupts per seconds between the card and the CPU.</div> <div class="MsoNormal"> I do not think it is a network congestion problem as intra-office communications as well as voicemail retrieval are always perfect. The Voip does not go over any routers, just a max of 2 switches with a 1GB trunk. This happens even off-hours when the network isn’t being used at
all.</div> <div class="MsoNormal"> There are never more than 2 people on the phone at the same time and it is definitely not an over-utilized processor.<br> </div> <div class="MsoNormal">I have trying to figure this out for 2 months on and off with no success any help is appreciated.<br> <br> <br> </div> Thanks<br><br><b><i>Andrew Shelton <andrew.shelton@stemnetworks.co.uk></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> <meta name="Generator" content="Microsoft Word 11 (filtered medium)"> <!--[if !mso]> <style> v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);} </style> <![endif]--><o:SmartTagType namespaceuri="urn:schemas-microsoft-com:office:smarttags" name="country-region">
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</o:SmartTagType></o:SmartTagType><div class="Section1"> <div class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">What codec are you using?<o:p></o:p></span></font></div> <div class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"><o:p> </o:p></span></font></div> <div class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">How many phone? What load is the server under?<o:p></o:p></span></font></div> <div class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"><o:p> </o:p></span></font></div> <div class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"><o:p> </o:p></span></font></div> <div
class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"><o:p> </o:p></span></font></div> <div> <div class="MsoNormal" style="text-align: center;" align="center"><font face="Times New Roman" size="3"><span style="font-size: 12pt;"> <hr tabindex="-1" align="center" size="2" width="100%"> </span></font></div> <div class="MsoNormal"><b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;"> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b><span style="font-weight: bold;">On Behalf Of </span></b>sdgesa gaeharth<br> <b><span style="font-weight: bold;">Sent:</span></b> 05 October 2006 13:22<br> <b><span style="font-weight: bold;">To:</span></b> asterisk-users@lists.digium.com<br> <b><span
style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute</span></font><o:p></o:p></div> </div> <div class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;"><o:p> </o:p></span></font></div> <div class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">1)Can anyone tell me how to do this on a Polycom 501?<br> <br> 2)Can you explain why you think this any why it ony happens on some calls?<br> <br> Thanks<br> <br> <b><i><span style="font-weight: bold; font-style: italic;">Andres <andres@telesip.net></span></i></b> wrote:<o:p></o:p></span></font></div> <div class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;"><br> ><br> ><br> > For about 20% of the calls to the outside world, the voice on the <br> > other end of an outside line is incredibly choppy.
Enough to where <br> > we have to hang up and call on a cell phone. It is always the same <br> > numbers that are choppy. The funny thing is, if I press mute while <br> > talking on a choppy call, the choppiness goes away completely.<br> ><br> > <br> ><br> Maybe you have silence suppression enabled on your phones. Try to <br> disable it and see if it helps.<br> <br> >------------------------------------------------------------------------<br> ><br> > <br> ><br> <br> <br> -- <br> Andres<br> Technical Support<br> http://www.telesip.net<br> <br> _______________________________________________<br> --Bandwidth and Colocation provided by Easynews.com --<br> <br> asterisk-users mailing list<br> To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<o:p></o:p></span></font></div> <div class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size:
12pt;"><o:p> </o:p></span></font></div> <div><font face="Times New Roman" size="3"><span style="font-size: 12pt;"> <o:p></o:p></span></font></div> <div class="MsoNormal" style="text-align: center;" align="center"><font face="Times New Roman" size="3"><span style="font-size: 12pt;"> <hr align="center" size="1" width="100%"> </span></font></div> <div class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">Yahoo! Messenger with Voice. <a href="http://us.rd.yahoo.com/mail_us/taglines/postman1/*http:/us.rd.yahoo.com/evt=39663/*http:/voice.yahoo.com">Make PC-to-Phone Calls</a> to the <st1:country-region w:st="on"><st1:place w:st="on">US</st1:place></st1:country-region> (and 30+ countries) for 2¢/min or less.<o:p></o:p></span></font></div> </div> _______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update
options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br></blockquote><br><p> 
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