[asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute

sdgesa gaeharth pollux1234567890 at yahoo.com
Fri Oct 6 10:19:54 MST 2006


Thanks for the reply...
  
  zapta.comf
    
    [channels]
    group = 1
    language=en
    context=incoming
    signalling=fxs_ks
    switchtype=national
    usecallerid=yes
    hidecallerid=no
    callwaiting=yes
    musiconhold=default
    usecallingpres=yes
    callwaitingcallerid=yes
    threewaycalling=yes
    transfer=yes
    canpark=yes
    cancallforward=yes
    callreturn=yes
    echocancel=yes
    echotraining=yes
    echocancelwhenbridged=yes
    rxgain=4
    txgain=-4
    channel => 1-4

  original Post:
  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
           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.
  
  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.
     
          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.
     
          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.
     
          There are never more than 2 people on the phone at the same time and it  is definitely not an over-utilized processor.
     
          I have trying to figure  this out for 2 months on and off with no success any help is appreciated.
  
  
     
      Thanks
  
  
Noah Miller <noahisaacmiller at gmail.com> wrote:  > Well I am using GSM as my main codec which seems to be > very nice


Polycom phones do not support GSM (GSM would not be necessary here
anyway, since all these phones are on a local LAN, so bandwidth does
not need to be conserved).


> You can also change some settings in the zapta and zaptel
> config.. to reduce
> echo and interference on the line..

This is the most important thing here - what does your zapata.conf look like?

Other things:
1. Update asterisk to a newer version.  There have been MANY bugs that
have been fixed since 1.2.4.
2. Update zaptel to a newer version.  Not much has changed for the TDM
cards since 1.2.7, but you should update anyway.

- Noah
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