Thanks for the reply...<br> <br> zapta.comf<br> <br> [channels]<br> group = 1<br> language=en<br> context=incoming<br> signalling=fxs_ks<br> switchtype=national<br> usecallerid=yes<br> hidecallerid=no<br> callwaiting=yes<br> musiconhold=default<br> usecallingpres=yes<br> callwaitingcallerid=yes<br> threewaycalling=yes<br> transfer=yes<br> canpark=yes<br> cancallforward=yes<br> callreturn=yes<br> echocancel=yes<br> echotraining=yes<br> echocancelwhenbridged=yes<br> rxgain=4<br> txgain=-4<br> channel => 1-4<br><br> original Post:<br> <div class="MsoNormal" style="margin-bottom: 12pt;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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</span></font></div> <div> <div class="MsoNormal"><font
face="Times New Roman" size="3"><span style="font-size: 12pt;">We are running asterisk-1.2.4 with zaptel-1.2.7 on Fedora Core 4-2.6.14-1.1656_FC4smp. It is 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. 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. Enough to where we have to hang up and call on a cell phone. It is always the same numbers that are choppy. The funny thing is, if I press mute while talking on a choppy call, the choppiness goes away completely.</span></font></div> </div> <div> <div
class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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.</span></font></div> </div> <div> <div class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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.</span></font></div> </div> <div> <div
class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">There are never more than 2 people on the phone at the same time and it is definitely not an over-utilized processor.</span></font></div> </div> <div> <div class="MsoNormal" style="margin-bottom: 12pt;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">I have trying to figure this out for 2 months on and off with no success any help is appreciated.<br> <br> </span></font></div> </div> <font face="Times New Roman" size="3"><span style="font-size: 12pt;">Thanks</span></font><br> <br> <br><b><i>Noah Miller <noahisaacmiller@gmail.com></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> > Well I am using GSM as my main codec which seems to be > very nice…<br><br>Polycom phones do not support GSM (GSM would not be necessary here<br>anyway, since all these
phones are on a local LAN, so bandwidth does<br>not need to be conserved).<br><br><br>> You can also change some settings in the zapta and zaptel<br>> config.. to reduce<br>> echo and interference on the line..<br><br>This is the most important thing here - what does your zapata.conf look like?<br><br>Other things:<br>1. Update asterisk to a newer version. There have been MANY bugs that<br>have been fixed since 1.2.4.<br>2. Update zaptel to a newer version. Not much has changed for the TDM<br>cards since 1.2.7, but you should update anyway.<br><br>- Noah<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<br></blockquote><br><p> 
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