Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
None of these steps have made a difference. Any other suggestions? Here is my original post:Can anyone help me to figure out why I can not write to a public share? I was able to join the domain without a problem. I can access the share from an xp box. I just can not write: "Access denied".thanksspecs:FC5 samba-common-3.0.23c-1.fc5 samba-3.0.23c-1.fc5smb.conf: [global] name resolve order = host disable netbios = yes workgroup = DOMAIN realm = DOMAIN.COM netbios name = NET preferred master = no server string = NET security = ADS encrypt passwords = yes log file = /var/log/samba/%m max log size = 50 winbind separator = + wins support = no idmap uid = 1-2 idmap gid = 1-2 winbind enum users = yes winbind enum groups = yes winbind use default domain = yes debug level = 2
[myshare] path = /usr/netshare writable = yes public = yes read >properties of the share: drwxrwxrwx 3 root root 4096 Sep 29 14:06 netshareRich Adamson [EMAIL PROTECTED] wrote: Noah Miller wrote: 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? zapta.comf switchtype=national This is not necessary in your case. It pertains to PRI lines, and not the POTS lines you have.echocancel=yes
echotraining=yes echocancelwhenbridged=yes You may want to turn each of these off, in turn, for testing, especially the "echocancewhenbridged". You can also tune the "echocancel" setting in terms of taps (a tap is one sample from the data stream per second). You can use the values: 16, 32, 64, 128, or 256 ('yes' just means 128).Might also try echotraining=800. That parameter causes the zaptel code to wait 800 milliseconds before pulsing the pstn line, and that pulse return is used to preload the software echo canceller to some reasonable starting point. Not usre if this will have any impact on your problem, but might be worth a try.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
Noah Miller wrote:
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?
zapta.comf
switchtype=national
This is not necessary in your case. It pertains to PRI lines, and not
the POTS lines you have.
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
You may want to turn each of these off, in turn, for testing,
especially the echocancewhenbridged.
You can also tune the echocancel setting in terms of taps (a tap is
one sample from the data stream per second). You can use the values:
16, 32, 64, 128, or 256 ('yes' just means 128).
Might also try echotraining=800. That parameter causes the zaptel code
to wait 800 milliseconds before pulsing the pstn line, and that pulse
return is used to preload the software echo canceller to some reasonable
starting point. Not usre if this will have any impact on your problem,
but might be worth a try.
___
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Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
Thanks for the reply...zapta.comf[channels]group = 1language=encontext=incomingsignalling=fxs_ksswitchtype=nationalusecallerid=yeshidecallerid=nocallwaiting=yesmusiconhold=defaultusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechotraining=yesechocancelwhenbridged=yesrxgain=4txgain=-4channel = 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 isnt 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. ThanksNoah Miller [EMAIL PROTECTED] 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 hereanyway, since all these phones are on a local LAN, so bandwidth doesnot 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 thathave been fixed since 1.2.4.2. Update zaptel to a newer version. Not much has changed for the TDMcards since 1.2.7, but you should update anyway.- Noah___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --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
Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
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?
zapta.comf
switchtype=national
This is not necessary in your case. It pertains to PRI lines, and not
the POTS lines you have.
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
You may want to turn each of these off, in turn, for testing,
especially the echocancewhenbridged.
You can also tune the echocancel setting in terms of taps (a tap is
one sample from the data stream per second). You can use the values:
16, 32, 64, 128, or 256 ('yes' just means 128).
- Noah
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RE: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sdgesa gaeharth Sent: 05 October 2006 14:38 To: asterisk-users@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 isnt 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 [EMAIL PROTECTED] wrote: What codec are you using? How many phone? What load is the server under? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sdgesa gaeharth Sent: 05 October 2006 13:22 To: asterisk-users@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 [EMAIL PROTECTED] 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. ___ --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 ___ --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
Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
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 ___ --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
