Displaying 20 results from an estimated 10000 matches similar to: "ldap password sync"
2006 Aug 27
1
migration strategy
I would like some advice on a migration strategy. Soon I will be migrating
from a old samba 2.2 server PDC server to a new samba 3 server (on new
hardware). I also wish to take this opportunity to rename the server which
I understand will change the SID.
What is the recommended method to achieve this without having to touch any
of the workstations?
Can I copy the old smbpasswd file to the new
2011 Apr 08
0
[LLVMdev] calling LLVM from C++ and passing C structs to work on?
I have written a parser and "engine" for a simple scripting language -
before I knew about LLVM.
I am thinking about whether it would be possible (and a good idea) to
rewrite it to use the LLVM.
A main issue that I am not sure if LLVM can provide is the following:
The script (which is called from C++ code) gets passed several C
structs that have been populated
by the
2008 May 17
1
smb_ldap_setup_connection failed with blank user Password
Hi,
i have a succesfully PDC Setup on a Novell/Suse OES1 SP2 Server with
eDirectory/LDAP and Samba Version 3.0.26a-0.7-1616-SUSE-SLES9.
Only Problem is, a User with a blank Password can't be authenticated on
a Domain Member Server (Users with Passwords doesn't have this Problem).
A Samba Level 10 Log show me this:
<snip>
[2008/05/14 16:25:48, 9]
2011 Dec 05
0
ANNOUNCEMENT: Call for Proposals for The R Series from Chapman & Hall/CRC
Chapman & Hall/CRC: The R Series
We are delighted to announce that our new series of books on R is up and running, with two books already published and another nine forthcoming (including three set to publish in 2012). We are keen to receive proposals for books covering all aspects of the development and application of R software. If you have an idea for a book, please contact one of the
2006 Jan 04
0
confusion about contexts - SER
Guess I got where your confusiion lies.... you DON'T set up the ALL contexts the users will have acces to in sip.conf, in this file you will only set 1 single context. All other contexts you want the users to have acces to, you will have to add them in the extensions.conf
So what you have to do is the following:
-user 2092, set it the createmenu context in sip .conf
- in extensions.conf
2006 Feb 01
3
XLite dtmf issue?
Hi,
I'm wondering if anyone has experienced an issue with the XLite
softphone and asterisk accepting dtmf? I can listen to my voicemail
perfectly from my hardphone. However when I dial the voicemail number
from my XLite softphone and enter the password at the voicemail prompt,
an error appears vm-incorrect and I get an "Unable to read password"
message on the asterisk console. Has
2009 Dec 07
0
Help required with NLME syntax
Hi there,
I am attempting to analyse data from a crossover study using a mixed
effect Emax model in R using nlme. I am having problems ascertaining how
to fit the model.
Subjects are randomly assigned to 1 of 4 treatments in a complete block
design. The response variable of interest is recorded once within each
period. So, subjects have a response for each period.
Specifically, I want to fit
2005 Feb 10
1
SER Asterisk Voicemail
Hi all,
I have SER and Asterisk set up together with ser handling user
registrations and asterisk providing voicemail services. When I ring
a phone and it doesnt answer after a designated amount of time, the
request is forwarded to asterisk, and I can leave a message.
Now, this may seem a ridiculous question but how can I listen to my
message afterwards? I have read about a solution by Java
2006 Jan 10
1
Asterisk voicemail support
Hi,
I was wondering if anyone has had a problem adding the 'delete' field to
the voicemail_users table. I have no problems adding other fields e.g.
alter table voicemail_users add column hidefromdir varchar(3) NOT NULL
default 'no';
However when I do
alter table voicemail_users add column delete varchar(3) NOT NULL
default 'no';
I get a message telling me that I
2005 Jan 11
1
asterisk one number service
I wonder does anyone have any thoughts or can give me some direction
on the following:
I have an asterisk testbed environment set up. My task is to make a
personal number service available whereby users would be given one
number (perhaps a voip number) and this number would enable them to
be reached via the pstn, pots, gsm etc....
Does anyone have ideas where I could start looking at sites to
2005 Mar 02
1
AD group membership limits?
Hi,
I'm running Samba 3.0.11 on RedHat ES 3 kernel version 2.4.21-15.0.4.ELsmp
and have a quick question about AD group membership limits
Am I right in assuming that Samba is limited by the group membership
parameters (ie NGROUP = 32) imposed by the Linux kernel? Is there any
workaround in Samba for this?
At the moment if a user is a member of more then 32 domain groups they
cannot access
2009 Jun 16
0
ANNOUNCEMENT: 20% discount on the most recent R books from Chapman & Hall/CRC!
Take advantage of a 20% discount on the most recent R books from Chapman & Hall/CRC!
We are pleased to offer our latest books on R at a 20% discount through our new website. To take advantage of this offer, simply visit http://www.crcpress.com/, choose your titles and insert code 281DW in the 'Promotion Code' field at checkout. Standard shipping is also free on all orders from
2005 Feb 14
2
FW: SER Asterisk Voicemail
Any more ideas on my below mail? If a user is registered with SER and
leaves a voicemail message with asterisk (by using rewritehostport
etc in ser.cfg), then how is the user supposed to listen to the
message afterwards? Is there any other way other than the MWI method??
Thnaksm
Aisling.
---- Original Message ----
From: ashling.odriscoll@cit.ie
To: asterisk-users@lists.digium.com
Subject: FW:
2006 Jan 05
1
Incoming PSTN Calls
Hi all,
I am having difficulty getting incoming PSTN calls working. I have set
up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc
My provider told me to change my sip.conf as follows
register => username:password@sip.blueface.ie/2093
; To receive incoming calls specify this block and
2005 Sep 06
1
Asterisk BT100 Password Issue
Hi,
I am getting the following error when I attempt to listen to voice
messages by dialing 9999 (I can hear nothing):
--Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack
--Playing 'vm-password' (language 'en')
WARNING: app_voicemail.c:4922 vm_authentication: Unable to read
password.
I read in previous posts that this may be to do with the dtmf
2005 Jan 24
4
ISP connection to the PSTN using Asterisk
Hi all,
Could someone let me know the most common way that an Internet ISP
would allow customers access to the PSTN?? Do they buy multiple fxo
cards such as the TDM400P and rent multiple lines from a larger
provider??
Would the best way be to connect to a third party voice/pstn
gateway?? Is that simply a matter of forwarding all sip traffic
destined for the pstn to another provider with a
2008 Dec 11
2
Simplex function in R
I have a set of linear equations and would like to find any feasible solution. A simplex solution works in Case 1 below, but not in Case 2. I would be grateful for any help.
Case 1: Find any feasible solution for the set of linear equations:
a + b + c = 5
a + b + 0c = 4
0a + b + c = 4
Solution - a feasible (and unique) solution is a=1, b=3, c=1.
The following R code
2003 Sep 19
0
/dev/random HP-UX 11.0
There is a /dev/random available now for HP-UX 11.0, see
http://www.josvisser.nl/hpux11-random/
Regards.
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2006 Nov 24
0
ssh bug on Reflection X.
Dear admin,
I am working in St George Bank, Australia. I have installed Openssh
4.0p1 and 4.5p1 on our Sun Solaris 8 unix servers. We experience some
bug when we launch a window on PC using ReflectionX by method Openssh to
any of our Solaris 8 servers, there is no update login and logout
information about users on /var/adm/wtmpx. Simply said when we using
Reflection X to login a Solaris 8 server
2005 Jan 15
0
configuring ser for *
Hi,
I currently have Asterisk running behind a linux router running nat.
Clients register with the public address and when the sip requests
reach the router, port forwarding is used to divert the traffic to *
i.e. all sip and rtp go to the asterisk box.
I now want to set up ser (so that i can have sip client with urls and
also to prevent the rtp stream going through asterisk). SER will also
be