similar to: Upgrading a jail

Full_Name: Kingsley Oteng-Amoako Version: 2.5.1 OS: Windows 5.1.2600 (Windows XP) Submission from: (NULL) (203.185.215.144) The col.names=false in the write.csv command does not work as documented. Attempting to write a vector to a csv file without column headers does not work as documented. The col.names=false feature on the write.table command does however work and it can thus be coerced

Hi, if we were to use shred to delete a file on a gluster volume, will the correct blocks be overwritten on the bricks? (still using Gluster 3.6.3 as have been too cautious to upgrade a mission critical live system). Cheers, Kingsley.

My Asterisk has started *frequently* segfaulting during IVR and voicemail activity -- it'll be playing a prompt gsm file to a caller and it'll just die (segfaulting, naturally dropping the caller) with no rhyme or reason. I've run gdb on four different core files and each one shows the reason for the segfault to be the following: #0 0x45df8b24 in Gsm_Short_Term_Synthesis_Filter

Hello, I'm looking forward to mining some new data six ways from Sunday with a great looking R application named "rattle". May I please have the benefit of this list's informed thoughts on how to debug an installation error? On Debian Linux's unstable release, I've done root$ aptitude install r-cran-rattle root$ aptitude install r-cran-rgtk2 user$ R

Hi, We're using Asterisk 13.17.0 with PJSIP 2.8 bundled. I've found an issue when Asterisk tries to make a SIP call out using auth, but has the wrong credentials and keeps getting returned a SIP 407, in this example to an OpenSIPs server requiring user auth. Basically this happens: 1. Asterisk sends plain INVITE to OpenSIPs 2. OpenSIPs responds with SIP 407 auth required with a

Is my understanding correct that the body() function currently can't return a function's body intact, in order, and as characters ready for editing? My testing and reading of body()'s help indicate that it can not. Here's what I'm seeing. Consider pasting 1+ and a function containing x^2 together to get 1+x^2 As you can see below, body() reports three

Hi, is this a rare problem? Cheers, Kingsley. On Tue, 2023-03-14 at 19:31 +0000, Kingsley Tart wrote: > Hi, > > using Gluster 9.2 on debian 11 I'm trying to set up geo replication. > I am following this guide: > > https://docs.gluster.org/en/main/Administrator-Guide/Geo-Replication/#password-less-ssh > > I have a volume called "ansible" which is only a

On 11/08/2017 11:36 PM, Kingsley Tart wrote: > Hi, > > if we were to use shred to delete a file on a gluster volume, will the > correct blocks be overwritten on the bricks? > > (still using Gluster 3.6.3 as have been too cautious to upgrade a > mission critical live system). When I strace `shred filename`, it just seems to write + fsync random values into the file based on

Hi, Does anyone know what it means when I've got an incoming fax routed through to iaxmodem+hylafax and then I see this in the asterisk log: DEBUG[18902] chan_dahdi.c: Detected digit 'f' This happens just after the initial fax negotiation has started and seems to correspond with the sending fax machine giving up. Googling hasn't helped me here :( -- Cheers, Kingsley.

On Wed, 2020-10-28 at 14:40 -0300, Joshua C. Colp wrote: > This is not yet fixed, but is being worked on. I have it as a > security issue currently out of caution (although I don't think we'll > treat it as one after further investigation). Right OK, thanks. Do you have any idea of the sort of timescale, and whether it'll be available as a patch that we can apply to our

Hi, We've recently hit an issue with Asterisk 18.8.0 where a call comes in via SIP (using pjsip) but it can take 5 seconds before starting to execute the dialplan. This was intermittent, but frequent (eg approx half of the calls). We have verbose logging on, but I didn't see any errors. Running asterisk -r -vvvv and then watching SIP traffic in another window showed the INVITE coming

Hi, I'm not sure whether this is possible but if it is, I'm sure someone on here might know ... Is it possible to use Monitor() to record a conversation[1], but make it start a new pair of wav files at intervals (eg every 15 minutes) if the calls go on for a long time? We already have this happening if the callers press a specific key sequence (which we've defined in features.conf)

Hi, using Gluster 9.2 on debian 11 I'm trying to set up geo replication. I am following this guide: https://docs.gluster.org/en/main/Administrator-Guide/Geo-Replication/#password-less-ssh I have a volume called "ansible" which is only a small volume and seemed like an ideal test case. Firstly, for a bit of feedback (this isn't my issue as I worked around it) I had this

Hi, I'm trying to receive faxes using hylafax / iaxmodem but I just can't get it to work. We're using Sangoma E1 cards and have calls coming in over PSTN. I've tried turning hardware echo cancellation off but it makes no difference. This is what I get in /var/spool/hylafax/log: [root at faxhost log]# cat c000000003 Jan 14 12:44:43.39: [ 3403]: SESSION BEGIN 000000003 18005551212

Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful.

Hi Is it possible to rsync a RedHat Linux 7.1 machine across a 56K dialup to a Win2000 machine? Thanks Wadeegh

Hi, What if some fail2ban magic could keep OpenSIPs response from hitting Asterisk after N attempts ? Le mer. 28 oct. 2020 à 18:32, Kingsley Tart - Barritel Ltd < kingsley.tart at barritel.com> a écrit : > Hi, > > We're using Asterisk 13.17.0 with PJSIP 2.8 bundled. > > I've found an issue when Asterisk tries to make a SIP call out using > auth, but has the wrong

HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call is always a ring group called '600', my problem is that after press 1 (but this problem is present also with press 2) before that the inbound call is transfer to extension pass 10/11 seconds ! In attach log file about incoming call. I use Trixbox with Asterisk-1.6.0.10. Thanks. ------ Salvatore.

Hi, We are trying to create a jail with FreeBSD 5.3 but it's fails with this error: cc -O -pipe -I/usr/obj/usr/src/i386/legacy/usr/include -c /usr/src/games/fortune/strfile/strfile.c make: don't know how to make /j/usr/lib/libc.a. Stop *** Error code 2 We are excuting those command in /usr/src: export D=/j make world DESTDIR=$D Are there any problem with FreeBSD 5.3? We have ever

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first