similar to: so whats next?

How can I diagnose why a resilver appears to be hanging at a certain percentage, seemingly doing nothing for quite a while, even though the HDD LED is lit up permanently (no apparent head seeking)? The drives in the pool are WD Raid Editions, thus have TLER and should time out on errors in just seconds. ZFS nor the syslog however were reporting any IO errors, so it weren''t the disks.

I am trying to put some smarts around a call to <%= error_messages_for %> so that when I first come into this page i.e. when my instance variable @target_list has not been defined yet, the page doesn''t break. I''m doing this: <% if @target_list? %> <%= error_messages_for ''target_list'' %> <% end %> I get a syntax error on line 1.

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

The REV I card shows up in the PCI table as: 02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or 02:05.0 Class 0280: e159:0001) Subsystem: Unknown device b119:0001 But the REV E/F shows up as: 02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface (or 02:0d.0 Class 0780: e159:0001) Subsystem: Unknown device b100:0003 One

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi Is there any tool out there wich allows you to use a flac file with ogg metadata as an the file of origin and converts its to an ogg.vorbis file without losing any of the metadata tag, I know there is a perl script wich does something similar but it keeps the metadata in separate files, basicly what I am asking is if there is any application that

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi Is there any tool out there wich allows you to use a flac file with ogg metadata as an the file of origin and converts its to an ogg.vorbis file without losing any of the metadata tag, I know there is a perl script wich does something similar but it keeps the metadata in separate files, basicly what I am asking is if there is any application that

When I use my phone to make VOIP call and another calls comes from POTS my phone rings to POTS caller. Why? Shouldn't it generate busy signal! -- #Joseph

Hi All, I am using PWLIB-1.6.6-1 and Openh323 1.13.5-1 and running a RH7.3 machine and I am unable to compile asterisk due to these errors. say.c: In function `powiedz': say.c:1633: parse error before `int' say.c:1636: `i1000E6' undeclared (first use in this function) say.c:1636: (Each undeclared identifier is reported only once say.c:1636: for each function it appears in.)

Hi, I had a FWD account set up with asterisk (using SIP) and it was working fine both ways. I switched to IAX2 and now I can't get incoming calls from FWD. People who call my FWD number get a "480 - user is not online" message without any traffic reaching my box. I can call FWD numbers fine over IAX2. It seems fwd isn't trying to place the call over IAX2 because it thinks

I am trying to get to the bottom of audio clicks, pops, dropouts with my Asterisk server. These occur even when the system is under minimal load (e.g. 1 Zap device in a queue being played music on hold) and occurs with both Zap and Sip devices so isn't network related. The audio problems occur at the same time on all channels and seems to be when Asterisk "gets busy" and uses

Okay, I''ve got Mandrake v9.2, and I just got my laptop working with a Dlink 514 router and Zyair B100 card. After getting it to work, I found that I had to stop shorewall before I could actually connect anywhere. So I went into /etc/shorewall and added wlan0 to interfaces and changed zones as well. This seems to work, I can connect now with shorewall running. Was this the right way to

again, under another subject: sorry, maybe an all too trivial question. But we have power data from J frequency spectra and to have the same range for the data of all our subjects, we just transformed them into % values, pseudo-code: power[i,j]=power[i,j]/sum(power[i,1:J]) of course, now we have a perfect linear relationship in our x design-matrix, since all power-values for each subject sum up

<copied from opensolaris-dicuss as this probably belongs here.> I kept on trying to migrate my pool with children (see previous threads) and had the (bad) idea to try the -d option on the receive part. The system reboots immediately. Here is the log in /var/adm/messages Feb 8 16:07:09 amber unix: [ID 836849 kern.notice] Feb 8 16:07:09 amber ^Mpanic[cpu1]/thread=ffffff014ba86e40: Feb 8

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hi, this is probably quite stupid but I have no clue what's wrong. Let's say I write the function hline <- function() { cat("\\hline \n") } and call hline() from within a Sweave chunk. Why is there no carriage return after the \hline in the resulting tex file? if I call hline() hline() in the chunk, then I get \hline \hline in the tex code without a linebreak in

Hi, i want to import data from .ods and .xls files in R on a linux system. Seems it was a faq in the past, but i found only solutions for Windows. Is there a handy solution for linux? The best would be something like mytab <-read.ods(...) Any hints? Thanks a lot for reading so far, -- Jonas Stein <news at jonasstein.de>

So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... [companyx-main-aa] exten => s,1,Background(companyx/companyx-main) exten => s,2,Background(silence/10) exten => s,3,Background(companyx/companyx-main) exten => s,4,Background(silence/10)

With a data frame (call it *d*) composed of 2000 individuals and *n*observations for each individual (thus *2000n* observations in total), I would like to generate *k* bootstrap samples with replacement from *d*. Amongst other variables, *d* has a numeric variable *id* taking on identical value for observations belonging to the same individual. Taking into consideration the panel nature of the

I have the user manual, I'll send it to your email tonight when I'll be in my home. I have an APA III-4FXO too, until today I can't put it to work with asterisk. Kind regards, Miguel Date: Fri, 03 Sep 2004 16:07:59 +1000 From: Jamie Carl <geek@j-code.net> Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual? To:

Folks, I've setup an IVR system that's been running fine (in production) for the last 6 months. It's primarily been used for North American calls. The problem I'm having is that for international calls, DTMF does not work. The tones just don't get detected. I tried a number of different codecs and options, but to no avail. I'm using IAX through my provider. The numbers