[asterisk-users] Extremely choppy sound on some of ourPOTSnetwork
calls; goes away with mute
Andrew Shelton
andrew.shelton at stemnetworks.co.uk
Thu Oct 5 07:09:02 MST 2006
Well I am using GSM as my main codec which seems to be very nice...
I would also suggest you looking at the load of you CPU... I know that asterisk is very processor hungry...
You can also change some settings in the zapta and zaptel config.. to reduce echo and interference on the line..
________________________________
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of sdgesa gaeharth
Sent: 05 October 2006 14:38
To: asterisk-users at lists.digium.com
Subject: RE: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
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
Andrew Shelton <andrew.shelton at stemnetworks.co.uk> wrote:
What codec are you using?
How many phone? What load is the server under?
________________________________
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of sdgesa gaeharth
Sent: 05 October 2006 13:22
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Extremely choppy sound on some of our POTSnetwork 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 at 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 <http://us.rd.yahoo.com/mail_us/taglines/postman1/*http:/us.rd.yahoo.com/evt=39663/*http:/voice.yahoo.com> to the US (and 30+ countries) for 2¢/min or less.
_______________________________________________
--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
________________________________
Get your email and more, right on the new Yahoo.com <http://us.rd.yahoo.com/evt=42973/*http:/www.yahoo.com/preview>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061005/6e7e8a95/attachment.htm
More information about the asterisk-users
mailing list