sdgesa gaeharth
2006-Oct-03 11:32 UTC
[asterisk-users] Extremely choppy sound on some of our POTS network calls; goes away with mute
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 --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2?/min or less. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061003/5fce1752/attachment.htm
Andres
2006-Oct-03 12:36 UTC
[asterisk-users] Extremely choppy sound on some of our POTS network calls; goes away with mute
> > > 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. > > >Maybe you have silence suppression enabled on your phones. Try to disable it and see if it helps.>------------------------------------------------------------------------ > > >-- Andres Technical Support http://www.telesip.net
sdgesa gaeharth
2006-Oct-05 05:22 UTC
[asterisk-users] Extremely choppy sound on some of our POTS network calls; goes away with mute
1)Can anyone tell me how to do this on a Polycom 501? 2)Can you explain why you think this any why it ony happens on some calls? Thanks Andres <andres@telesip.net> wrote:> > > 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. > > >Maybe you have silence suppression enabled on your phones. Try to disable it and see if it helps.>------------------------------------------------------------------------ > > >-- Andres Technical Support http://www.telesip.net _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2?/min or less. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061005/4371aedc/attachment.htm