similar to: add machine script

Hello, I'm using samba 2.2.3a in debian woody. I've set up the debian server as a PDC, and clients can join and log in fine. What I'm now looking for is how to set up groups for my users. When I try to set the security for a file in windows, I see the users on the linux server, but no groups. How do I map between unix group and samba groups? I've seen mention of the

Hi there, Sorry to possibly rehash stuff that's already been answered, I'm new to the list. I've recently installed 3.0alpha23-0 because of it's superior group mapping (over 2.2). I'm running a PDC on Debian Woody. I'm also using the tdb backend as ldap is too big a bite to chew for me for now. Things are working well for basic shares, but I want to get into setting

Dear R experts, I fitted data (length == m) with an external library (that's a non-linear problem), and, I've got the final parameter vector (length == n) and Jacobian matrix (dim == c(m, n)). Now, I want to analyse the standard errors of the parameters correctly. How can I do it? Where can I read about it else? Thanks! -- WBR, Timur.

Am 23.06.2020 15:43, schrieb Marek Greško: Hi >> Do you mean "my Linux-Box ignores ICMP packet unreachable" or >> "Deutsche >> Telekom ignores them"? > > I meant DT, but this was a speculation. I did not say they do. I > consider it highly improbable. Then I was asking whether you do. As > per configuration you sent you are not blocking icmp

Hi all, We followed some books and sample codes and did some EMM estimation, only to find it won't be able to generate forecast. This is because in the stochastic volatility models we are estimating, the volatilities are latent variables, and we want to forecast 1-step ahead or h-step ahead volatilities. So it is nice to have the system estimated, but we couldn't get it to forecast at

Am 23.06.2020 16:22, schrieb Marek Greško: > It seems your problems lie in something other. Most probably it is not > mtu problem. All my suspections are contradicted. If it is true you > have inter vlan voice quality problems, it is definitely something > different. Formerly I assumed you were trying only LTE vs LAN using > internet. I'm not sure what you mean with the last

Hello everybody, I'm experimenting with samba 3.0 but something is not very clear to me, especially the user entries in the OpenLDAP server. On my samba server, I configured NSS_LDAP to lookup passwd, group and shadow entries. Now I have to add some entries to my LDAP: uid=Administrator and uid=nobody. According to http://www.unav.es/cti/ldap-smb/smb-ldap-3-howto.html a user with uidNumber 0

> Mr. Andrew Tridgell: > > I have made a gallant attempt to install and make use of the Samba 2.0.7 > suite of disk-sharing utility software > on my Sun E-3000 running Solaris 2.5.1. (You will probably think my > attempt was not gallant enough > since I'm not currently able to use the package. Sorry...) > > While we are a City, and have a substantial desktop

Sorry, I forgot to write that the SIP peer must keep ringing while the announcement is being played. Le 22/08/2016 ? 17:42, John Kiniston a ?crit : > This seems like the obvious answer but maybe I'm misunderstanding the > question. > > exten => s,1,Dial(SIP/alice,20) > same => n,Playback(myannouncement) > same => n,NoOP(Whatever else you want to do goes

It seems your problems lie in something other. Most probably it is not mtu problem. All my suspections are contradicted. If it is true you have inter vlan voice quality problems, it is definitely something different. Formerly I assumed you were trying only LTE vs LAN using internet. Marek 2020-06-23 15:50 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>: > Am 23.06.2020 15:43,

I interchanged LAN and LTE in the sentence. Do you have some kind of NAT in fron of asterisk? Or is your asterisk having public IP? Could you share sip.conf (without passwords)? One LAN client, one LTE and general section. Marek 2020-06-23 16:29 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>: > Am 23.06.2020 16:22, schrieb Marek Greško: >> It seems your problems lie in

It wouldn''t be proper to start my first post here without congratulations and thanks to the ZFS team for such an impressive piece of work. Anyway, on to my query. I''ve been trying out ZFS, with a particular focus in reducing latency in a specific application. This application has a fair amount of random writing going on in the background (which, of course, ZFS will make

That's just what I'm using, John. But I'm getting (eg) [Nov 4 21:46:16] ERROR[1676][C-00000003]: func_env.c:449 file2format: Cannot open '/home/logs/anonymous.txt': No such file or directory [Nov 4 21:46:16] ERROR[1676][C-00000003]: func_env.c:949 file_write: File '/home/logs/anonymous.txt' not in line format Asterisk is running as root (yeah, I know!), and has

I recently moved my openSUSE 11.4 system from the proprietary drivers to nouveau. After much grief due to the sometimes conflicting instruction on the Internet and applications that placed out of date shared objects (libraries) in /usr/local/lib which is searched first by ld.so, I have it working on a 19" wide screen LCD monitor at 1440x900. However, there is a distinct and periodic flicker

Hi, 1. How do you then, synced then unread message presence with custom device status ? From an external program ? When a user leaves VoiceMailMan application ? Using externnotify ? 2. What is MWI:101 at default expression for (see [2] ? Cheers [2] https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+button+to+Monitor+a+Voicemail+Box 2017-11-21 17:58 GMT+01:00 John Kiniston <johnkiniston at

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial, It supports playing > an announcement. >

I'm implementing a SBC with my Asterisk PBX but the keeps disabling the trunk group I've configured and I think it may be because Asterisk is returning a 4r04 to the OPTIONS. I've created a test context and have put in a wildcard pattern match to try and catch those options but it doesn't seem to work. Is there a way to have asterisk respond with an 200 OK instead of a 404? --

More testing. Test test test. :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

That is the general idea. But how do i make it work? is there somewhere ready? On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkiniston at gmail.com> wrote: > Define your *72 and *73 extensions in your internal context, Have them set > a value in the ASTDB that you then check when dialing your handsets. > > The same can be done for call forwarding, store a number in the

John, >From looking at the wiki won't STRFIME just give me what I need based on the unix time that I put in? What I am actually looking to do is convert over from 12 hour format to 24 (unless strftime does just that and I don't kow what am I am doing?). On Thu, Feb 13, 2020 at 12:03 PM John Kiniston <johnkiniston at gmail.com> wrote: > Try using the STRFIME function