[Asterisk-Dev] Re: IRQ misses and crackles during disk activity
John Matthews
john.matthews at crossml.com
Mon Jun 14 21:01:26 MST 2004
For the record we fixed this problem by changing to a 3Ware escalade disk
adapter. No more crackles, patlooptest doesn't show any errors and also disk
access is a lot quicker.
John Matthews
CROSSML
Phone: 1300 305 110
-----Original Message-----
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Today's Topics:
1. Re: IRQ misses and crackles during disk activity (Steven Critchfield)
2. handsfree application and asterisk (Andreas Bayer)
3. Re: Centralized voicemail (Rob Gagnon)
4. Re: IRQ misses and crackles during disk activity (Peter Svensson)
5. Remote-Party-ID patch (=?iso-8859-1?Q?Niklas_=D6gren?=)
6. RE: Making libiax2 speak TCP (through udp tunnelin
g) (Whisker, Peter)
--__--__--
Message: 1
Subject: Re: [Asterisk-Dev] IRQ misses and crackles during disk activity
From: Steven Critchfield <critch at basesys.com>
To: asterisk-dev at lists.digium.com
Date: Sun, 13 Jun 2004 21:25:05 -0500
Reply-To: asterisk-dev at lists.digium.com
On Sun, 2004-06-13 at 21:09, John Matthews wrote:
>
>
> I am using an E100P with all the latest asterisk and zaptel versions
> (0.9.1). I carefully ensured that my E100P wasnÿt sharing interrupts
> with any other devices in my system. Problem is that I get the
> occasional crackle during phone conversations. Looking deeper the
> crackles are always associated with IRQ misses which seem to occur
> when data is being read/written to the disk. Crackles always occur
> when the disk active LED on my PC is blinking. Running patlooptest
> (with a loopback adapter fitted to the card) shows essentially the
> same thing ie) data errors at times when my disk is active. It
> occurred to me that the fault may lie as much with the disk drivers as
> with the E100P. If they were disabling interrupts for ´lengthy¡
> periods then it wouldnÿt be surprising that the E100P was not able to
> service its interrupts in a timely manner. My system is fitted with a
> standard IDE disk. If anyone has a system that works without crackles
> and IRQ misses please could you let me know what type of disks you are
> using.
Use hdparm to look over your hard drive settings. You need 32 bit access
and you need dma turned on. This will let your hard drive activity be
more efficient and not take up as much time.
Usually I use "hdparm -c1 -d1 /dev/hda" if I need to pursuade the kernel
of the settings.
--
Steven Critchfield <critch at basesys.com>
--__--__--
Message: 2
From: Andreas Bayer <angel_azrael at gmx.de>
To: vogl+hfp at soft.uni-linz.ac.at
Date: Mon, 14 Jun 2004 04:38:27 +0200
Cc: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] handsfree application and asterisk
Reply-To: asterisk-dev at lists.digium.com
Hi,
i have heard about you little handsfree application.
I would like to know if there are any plan to implement it in a voip gateway
like asterisk?
Bye
--__--__--
Message: 3
From: "Rob Gagnon" <rob at networkip.net>
To: <asterisk-dev at lists.digium.com>
Subject: Re: [Asterisk-Dev] Centralized voicemail
Date: Sun, 13 Jun 2004 23:58:39 -0500
Organization: Network IP
Reply-To: asterisk-dev at lists.digium.com
I will be coding a fix in ast_data for this sometime on Monday. The
work-around for now is to make sure to always name your context when calling
"VoiceMailMain()" in extensions.conf. [IE: "VoiceMailMain(@default)" ]
There is an issue leaving a NULL string pointer when context is missing from
a message in ast_data. The solution will be fairly trivial, but I cannot
currently access the server in my office to fix it. (seems a power outage
there is preventing that)
Thanks Soren, for reporting this issue. It will definitely help in making
ast_data even more stable than it is now.
Rob
----- Original Message -----
From: "Rob Gagnon" <rob at networkip.net>
To: <asterisk-dev at lists.digium.com>
Sent: Sunday, June 13, 2004 5:58 PM
Subject: Re: [Asterisk-Dev] Centralized voicemail
> So everyone knows, Soren and I are investigating this issue outside of the
> mailing list. (just to save some un-wanted copies of logs for
everyone)....
>
> I will post the results when, or if anything is found.
>
> Rob
>
> ----- Original Message -----
> From: "Soren Rathje" <asterisk at lolle.org>
> To: <asterisk-dev at lists.digium.com>
> Sent: Sunday, June 13, 2004 2:23 PM
> Subject: Re: [Asterisk-Dev] Centralized voicemail
>
>
> Rob,
>
> I don't know if you received the data I sent the other day, but I repeated
> the exercise after disabling NTLS as I read somewhere it could cause
> problems with no difference....
>
> If you need hands-on access let me know off-list on soren (at) lolle.org.
>
> -- Soren
>
> ----- Original Message -----
> From: "Rob Gagnon" <rob at networkip.net>
> To: <asterisk-dev at lists.digium.com>
> Sent: Friday, June 11, 2004 8:58 PM
> Subject: Re: [Asterisk-Dev] Centralized voicemail
>
>
> > Soren,
> >
> > Information about the methods followed to get to the error, as well as
any
> > log messages prior to it would be nice.
> >
> > Also, if you know how to use GDB, that would be helpful to get the last
> few
> > things that happened.
> >
> > Rob
> >
> > ----- Original Message -----
> > From: "Soren Rathje" <asterisk at lolle.org>
> > To: <asterisk-dev at lists.digium.com>
> > Sent: Friday, June 11, 2004 12:17 PM
> > Subject: Re: [Asterisk-Dev] Centralized voicemail
> >
> >
> >
> > ----- Original Message -----
> > From: "brian k. west" <brian at bkw.org>
> > To: <asterisk-dev at lists.digium.com>
> > Sent: Sunday, May 09, 2004 4:59 AM
> > Subject: Re: [Asterisk-Dev] Centralized voicemail
> >
> >
> > > its ast_data now.. its been brought into the core of asterisk.
> > >
> > > bkw
> > >
> >
> > Brian, I get a segfault on VoiceMail with ast_data version 133, 134,
136,
> > 141 with current HEAD 11-06-04.
> >
> > Plain vanilla FC1, MySQL, PostgreSQL, unixODBC.
> >
> > segfault reproduceable every time on MySQL, PgSQL and ODBC(PgSQL).
> >
> > What do you need to debug, when it comes to Linux I'm still a virgin..
:-)
> >
> > -- Soren
> >
> > _______________________________________________
> > Asterisk-Dev mailing list
> > Asterisk-Dev at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> > _______________________________________________
> > Asterisk-Dev mailing list
> > Asterisk-Dev at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
>
> _______________________________________________
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--__--__--
Message: 4
Date: Mon, 14 Jun 2004 07:41:44 +0200 (CEST)
From: Peter Svensson <psvasterisk at psv.nu>
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] IRQ misses and crackles during disk activity
Reply-To: asterisk-dev at lists.digium.com
On Mon, 14 Jun 2004, John Matthews wrote:
> I am using an E100P with all the latest asterisk and zaptel versions
> (0.9.1). I carefully ensured that my E100P wasn't sharing interrupts with
> any other devices in my system. Problem is that I get the occasional
crackle
> during phone conversations. Looking deeper the crackles are always
> associated with IRQ misses which seem to occur when data is being
> read/written to the disk. Crackles always occur when the disk active LED
on
> my PC is blinking. Running patlooptest (with a loopback adapter fitted to
> the card) shows essentially the same thing ie) data errors at times when
my
> disk is active. It occurred to me that the fault may lie as much with the
> disk drivers as with the E100P. If they were disabling interrupts for
> "lengthy" periods then it wouldn't be surprising that the E100P was not
able
> to service its interrupts in a timely manner. My system is fitted with a
> standard IDE disk. If anyone has a system that works without crackles and
> IRQ misses please could you let me know what type of disks you are using.
The ide driver in Linux is run with other interupts disabled by default.
Use "hdparm -u1 /dev/hdX" to enable other interrupts while serving the ide
drives.
Peter
--
Peter Svensson ! Pgp key available by finger, fingerprint:
<petersv at psv.nu> ! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF
------------------------------------------------------------------------
Remember, Luke, your source will be with you... always...
--__--__--
Message: 5
Date: Mon, 14 Jun 2004 11:22:23 +0200 (CEST)
From: =?iso-8859-1?Q?Niklas_=D6gren?= <list_ast at nod.se>
To: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] Remote-Party-ID patch
Reply-To: asterisk-dev at lists.digium.com
Hi!
I just submitted http://bugs.digium.com/bug_view_page.php?bug_id=0001841
to address a feature request of being able to set the correct A number
when calling from SIP, through Asterisk as a gateway to PSTN using ISDN.
It is taking the Remote-Party-ID header and if privacy is not set to
"full" or "uri" in that header, the number is taken and set as callerid
number instead of the number found in "from" header.
If there are sip routers that are the only ones that are allowed to send
sip requests to Asterisk, they can make sure that the remote-party-id
header is valid with their configs, and the header is valid to use in
asterisk.
I also changed chan_modem and chan_modem_i4l to take care of the callerid,
check with a "valid-list" in modem.conf, and set that number as outgoing
MSN.
Please review and merge if it is working as expected. (it works for me..
:) )
/Niklas
--__--__--
Message: 6
From: "Whisker, Peter" <Peter.Whisker at logicacmg.com>
To: "'Asterisk-Dev'" <asterisk-dev at lists.digium.com>
Subject: RE: [Asterisk-Dev] Making libiax2 speak TCP (through udp tunnelin
g)
Date: Mon, 14 Jun 2004 14:23:41 +0100
Reply-To: asterisk-dev at lists.digium.com
I have had some success bypassing a firewall using UDP tunelling through TCP
to a machine outside the firewall using the program Zebedee
(http://www.winton.org.uk/zebedee/index.html).
The zebedee client and server setup files are as follows:
Server:
======
E:\Zebedee>type server.zbd
#
verbosity 2 # Slightly more than basic messages
detached true
server true # Yes, it's a server!
ipmode both # Operate in mixed TCP/UDP mode
compression zlib:9 # Allow maximum zlib compression
keylength 256 # Allow keys up to 256 bits
keylifetime 36000 # Shared keys last 10 hours
maxbufsize 16383 # Allow maximum possible buffer size
logfile './server.log'
keygenlevel 2 # Generate maximum strength private keys
checksumlevel 7 # Allow maximum strength checksums
minchecksumlevel 0 # Allow no checksums if client requests
checkidfile './clients.id'
serverport 750
target iax2.fwdnet.net
target localhost
Client:
======
/etc# cat zeb_cli.conf
include "/etc/zebedee.key"
multiuse true
detached true
serverhost 146.101.4.28
serverport 750
ipmode both
compression zlib:0
checksumlevel 0
maxbufsize 1024
tunnel 4569/udp:iax2.fwdnet.net:4569
The client and server use TCP port 750 which is open outbound on my
firewall. The Client sets the Max buffer size to 1024 (a bit bigger than we
need but not too big otherwise latency goes up) - you can go down to < 800
but then start to have problems. The Asterisk server can not be the same as
the Zebedee client otherwise both want to bind 4569/udp. Use the
zebedee_client at your.net as the address of the server (iax2.fwdnet.net in
this case) in iax.conf. You can't connect to more than one IAX server doing
it this way as something (Asterisk? Zebedee?) gets confused, even using
different UDP ports. It seems to try and register everything on one server.
Zebedee works well in Linux & Windows but get the latest development
version.
This setup works for me for now while I get my firewall opened.
Peter
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