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List Info
Thread: Reliability
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| Reliability |
  Canada |
2008-02-07 08:34:48 |
I just set up Asterisk 1.4.11 on a FreeBSD 6.2 system, and
have been testing
it for the last month or so.
I have a single FXO interface installed to provide timing.
The server is a
Compaq DL380G2 with Dual 1GHz Xeon chips and about a Gig of
ram.
When I was running Linux, it was a pain to administer, but
didn't give me any
grief. Now, I'm having calls not hungup, unexplained
asterisk core dumps,
unsuccessful calls, and calls dropped when they are on
hold.
All channels are SIP only. No g729, no transcoding, just
pure SIP.
I haven't seen a load average greater than 0.3 on this
machine. I usually
don't get more than 16 channels at once, and see about
20,000 calls per
month.
I'm thinking I should go back to Linux, but I really would
prefer BSD.
Am I doing something wrong here, or is the BSD port just not
as reliable?
Thanks
-Tim
--
Tim St. Pierre
IP telephony specialist
sip://5101 communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
tim communicatefreely.net
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om--
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| Re: Reliability |

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2008-02-07 08:51:28 |
On Thu, 7 Feb 2008, Tim St. Pierre wrote:
> I just set up Asterisk 1.4.11 on a FreeBSD 6.2 system,
and have been testing
> it for the last month or so.
> I have a single FXO interface installed to provide
timing. The server is a
> Compaq DL380G2 with Dual 1GHz Xeon chips and about a
Gig of ram.
>
> When I was running Linux, it was a pain to administer,
but didn't give me any
> grief. Now, I'm having calls not hungup, unexplained
asterisk core dumps,
> unsuccessful calls, and calls dropped when they are on
hold.
>
> All channels are SIP only. No g729, no transcoding,
just pure SIP.
> I haven't seen a load average greater than 0.3 on this
machine. I usually
> don't get more than 16 channels at once, and see about
20,000 calls per
> month.
> I'm thinking I should go back to Linux, but I really
would prefer BSD.
>
> Am I doing something wrong here, or is the BSD port
just not as reliable?
Tim, I've been running asterisk on FreeBSD for a bit over 3
years now.
The current setup is a 750MHz PIII with 256MB ram and 6.2.
The previous
was on a dual 550 Xeon running 4.7. The current one has a
single FXO
in it (X101P). Both machines are Dell servers.
I've never had a core dump and the only problems I've had
were minor
config issues (the kind that are staring you in the face and
you still
don't see them) and changes that my voip provider made that
I didn't
catch.
The asterisk version I'm currently running is 1.4.0 (one day
I'll get
energetic and upgrade it).
Have you analyzed the core dumps to see why or where it
fails?
Vince.
--
Michigan VHF Corp. http://www.nobucks.net/
http://www.coloser.com/
Bringing the world a little coloser to your
business
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om--
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| Re: Reliability |
  Canada |
2008-02-07 09:09:08 |
I'm not much of a programmer, so I haven't taken a crack at
it a core dump.
I was trying some creative logging to find patterns though.
It looked like it
would get panicky when Meetme rooms closed, or if I did a
pickup through a
local channel (pickup was trying to re-invite, even though
canreinvite=no),
so I used to direct it through a local channel to get around
that.
Putting the Zaptel card in seemed to help a lot.
I haven't had a core dump since I removed the pickup through
local, but I'm
still getting erratic behaviour at times.
If I do a sip show channels, I get 100 or more channels,
mostly SUBSCRIBE. I
could have 40 for one extension, that only has 7
subscriptions on it.
I have the entire overseas rate table listed as part of the
dialplan, which is
fairly large. I would like to move that to a database
lookup, but I haven't
figured out how to do the pattern matching with a variable
length key (some
country codes are longer than others).
I have 1.4.17 installed on a fresh DL360 (same specs, just
fewer drive slots),
and I am debating whether to move everyone to that machine
(It would be phone
only), or to rebuild it with Linux.
If everyone else out there has been doing high volume
production setups
without issue, I would love to stay with BSD. As far as
asking for help with
crashes and strange things, what info is useful to post?
Thanks
-Tim
On Thursday 07 February 2008 09:51, Vince Vielhaber wrote:
> On Thu, 7 Feb 2008, Tim St. Pierre wrote:
> > I just set up Asterisk 1.4.11 on a FreeBSD 6.2
system, and have been
> > testing it for the last month or so.
> > I have a single FXO interface installed to provide
timing. The server is
> > a Compaq DL380G2 with Dual 1GHz Xeon chips and
about a Gig of ram.
> >
> > When I was running Linux, it was a pain to
administer, but didn't give me
> > any grief. Now, I'm having calls not hungup,
unexplained asterisk core
> > dumps, unsuccessful calls, and calls dropped when
they are on hold.
> >
> > All channels are SIP only. No g729, no
transcoding, just pure SIP.
> > I haven't seen a load average greater than 0.3 on
this machine. I
> > usually don't get more than 16 channels at once,
and see about 20,000
> > calls per month.
> > I'm thinking I should go back to Linux, but I
really would prefer BSD.
> >
> > Am I doing something wrong here, or is the BSD
port just not as reliable?
>
> Tim, I've been running asterisk on FreeBSD for a bit
over 3 years now.
> The current setup is a 750MHz PIII with 256MB ram and
6.2. The previous
> was on a dual 550 Xeon running 4.7. The current one
has a single FXO
> in it (X101P). Both machines are Dell servers.
>
> I've never had a core dump and the only problems I've
had were minor
> config issues (the kind that are staring you in the
face and you still
> don't see them) and changes that my voip provider made
that I didn't
> catch.
>
> The asterisk version I'm currently running is 1.4.0
(one day I'll get
> energetic and upgrade it).
>
> Have you analyzed the core dumps to see why or where it
fails?
>
> Vince.
--
Tim St. Pierre
IP telephony specialist
sip://5101 communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
tim communicatefreely.net
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
Asterisk-BSD mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-bsd
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| Re: Reliability |
  United States |
2008-02-07 10:41:21 |
On February 7, 2008 07:09:08 am Tim St. Pierre wrote:
> I'm not much of a programmer, so I haven't taken a
crack at it a core dump.
>
> I was trying some creative logging to find patterns
though. It looked like
> it would get panicky when Meetme rooms closed, or if I
did a pickup through
> a local channel (pickup was trying to re-invite, even
though
> canreinvite=no), so I used to direct it through a local
channel to get
> around that.
>
> Putting the Zaptel card in seemed to help a lot.
>
> I haven't had a core dump since I removed the pickup
through local, but I'm
> still getting erratic behaviour at times.
>
> If I do a sip show channels, I get 100 or more
channels, mostly SUBSCRIBE.
> I could have 40 for one extension, that only has 7
subscriptions on it.
>
> I have the entire overseas rate table listed as part of
the dialplan, which
> is fairly large. I would like to move that to a
database lookup, but I
> haven't figured out how to do the pattern matching with
a variable length
> key (some country codes are longer than others).
>
> I have 1.4.17 installed on a fresh DL360 (same specs,
just fewer drive
> slots), and I am debating whether to move everyone to
that machine (It
> would be phone only), or to rebuild it with Linux.
>
> If everyone else out there has been doing high volume
production setups
> without issue, I would love to stay with BSD. As far
as asking for help
> with crashes and strange things, what info is useful to
post?
>
> Thanks
>
> -Tim
>
> On Thursday 07 February 2008 09:51, Vince Vielhaber
wrote:
> > On Thu, 7 Feb 2008, Tim St. Pierre wrote:
> > > I just set up Asterisk 1.4.11 on a FreeBSD
6.2 system, and have been
> > > testing it for the last month or so.
> > > I have a single FXO interface installed to
provide timing. The server
> > > is a Compaq DL380G2 with Dual 1GHz Xeon chips
and about a Gig of ram.
> > >
> > > When I was running Linux, it was a pain to
administer, but didn't give
> > > me any grief. Now, I'm having calls not
hungup, unexplained asterisk
> > > core dumps, unsuccessful calls, and calls
dropped when they are on
> > > hold.
> > >
> > > All channels are SIP only. No g729, no
transcoding, just pure SIP.
> > > I haven't seen a load average greater than
0.3 on this machine. I
> > > usually don't get more than 16 channels at
once, and see about 20,000
> > > calls per month.
> > > I'm thinking I should go back to Linux, but I
really would prefer BSD.
> > >
> > > Am I doing something wrong here, or is the
BSD port just not as
> > > reliable?
> >
> > Tim, I've been running asterisk on FreeBSD for a
bit over 3 years now.
> > The current setup is a 750MHz PIII with 256MB ram
and 6.2. The previous
> > was on a dual 550 Xeon running 4.7. The current
one has a single FXO
> > in it (X101P). Both machines are Dell servers.
> >
> > I've never had a core dump and the only problems
I've had were minor
> > config issues (the kind that are staring you in
the face and you still
> > don't see them) and changes that my voip provider
made that I didn't
> > catch.
> >
> > The asterisk version I'm currently running is
1.4.0 (one day I'll get
> > energetic and upgrade it).
> >
> > Have you analyzed the core dumps to see why or
where it fails?
> >
> > Vince.
ook for timing to work correctly with ztdummy. you have to
do a few things.
1 cd /usr/local/etc/rc.d and edit the zaptel script. as it
is missing ztdummy
in the module loads
on the end of the first line of modules before the "
put ztdummy.ko and after
the first " on the next line.
then you have to build a new kernel with the setting
"options HZ=1000
once you have done these 2 steps ztdummy should work.
the other issue is you need to streamline your install by
disabling modules
you dont need.Look at the modules.conf and learn to disable
modules. this
will help lower some resourcec. also if you dont need it do
not install h323
and postgress/radious server. its all about stream lining
your build.
if you need help with this let me know I can walk you threw
this.
--
Welcome to the World. An the World gets smaller.
_______________________________________________
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om--
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| Re: Reliability |
  United States |
2008-02-07 10:52:38 |
Tim St. Pierre wrote:
> I just set up Asterisk 1.4.11 on a FreeBSD 6.2 system,
and have been testing
> it for the last month or so.
> I have a single FXO interface installed to provide
timing. The server is a
> Compaq DL380G2 with Dual 1GHz Xeon chips and about a
Gig of ram.
>
> When I was running Linux, it was a pain to administer,
but didn't give me any
> grief. Now, I'm having calls not hungup, unexplained
asterisk core dumps,
> unsuccessful calls, and calls dropped when they are on
hold.
>
> All channels are SIP only. No g729, no transcoding,
just pure SIP.
> I haven't seen a load average greater than 0.3 on this
machine. I usually
> don't get more than 16 channels at once, and see about
20,000 calls per
> month.
> I'm thinking I should go back to Linux, but I really
would prefer BSD.
>
> Am I doing something wrong here, or is the BSD port
just not as reliable?
>
> Thanks
>
> -Tim
>
Please let us know if you find a cause resolution to some of
your
problems. I'm also having some trouble with calls not hung
up, both SIP
to SIP and SIP to Zap channels. I haven't experienced your
other
problems although we do only about a quarter of your volume.
My other
problem has been with asterisk dropping calls because it
attempts
changing to g.722 even though it's denied in sip.conf and
SIP phones as
an option. That's a nasty one but fairly infrequent.
--
Adam Vandemore
Systems Administrator
IMED Mobility
_______________________________________________
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om--
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To UNSUBSCRIBE or update options visit:
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