Displaying 20 results from an estimated 1000 matches similar to: "Standalone server providing authenticated services to AD clients?"
2013 Jan 16
1
Libvirt daemon is not getting started
Hi,
Earlier I was having libvirt-0.7.5 but now I have installed 0.9.8 version
of libvirt now.
I am trying to start libvirt daemon using the command below but daemon it
is not getting started and the PID file is not getting created. Tried
/usr/loval/sbin/libvirtd -d option too. In that case I am not getting any
error but PID file is not getting created in this case too.
2002 Aug 09
0
Automation of public/private key generation
Hi all,
I wrote a small script (developed and testet on Solaris 8), which
automates the generation and installation of the steps needed to put
keys in place. I you are interested to take it, feel free to do it.
--
*** Freundliche Gruesse **** Best regards ***
Anton Burkhalter
Dipl. El. Ing. HTL
Mobile:+41(0)78 844-0290
mailto:anton.burkhalter at gmx.net
2018 Apr 17
0
Bug: Dovecot index loosing sync with FTS despite "fts_autoindex = yes"
Le 17/04/2018 ? 14:18, kfx a ?crit?:
> dovecot 2.2.34
> solr 7.2
>
> I only see new messages after typing on the server "doveadm fts rescan
> -u username" though I've followed the wiki and added "fts_autoindex =
> yes" in 90-plugin.conf . Subsequent search for the same pattern always
> gives the same result, ignoring new emails with that particular
2011 Jul 29
0
Asterisk SIP authentication against [section] instead of username
Hello,
Asterisk seems to try to authenticate incoming INVITE based on the [section]
in sip.conf and not the username specified.
I just removed the "insecure" option from my sip.conf requesting every
connection to be authenticated. I added the match_auth_username=yes in the
[general] section for extra security. To make it work, I have to use the
same [section] identifier as username.
2004 Apr 06
1
SIP phone registering problem
I am clearly doing something ridiculously wrong.
Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are
unable to register. They keep trying and then time out.
With the sip debug on in Asterisk nothing is logged.
Here is the trace from one of the phones (kphone):
(192.168.100.13 is kphone, 192.168.100.3 is Asterisk)
sipclient: sending: 21:47:45.454
2006 Jan 19
0
Incoming fax on voipbuster
Hello,
I'm trying to receive a fax to my inbound number from voipbuster.
Asterisk receives the call and starts the rxfax application successful,
but then nothing happens. The calling party is still hearing a ringing
tone, or sometimes nothing. Voicecalls are working correct and without
problems.
For testing I've add a local number (300) to the dialplan. When I call
this number
2004 Jul 13
2
IAX2 calls through IAXTEL.com
I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls
through. IAXTEL.com gave me a number (example) of 700-555-6226. I have
made the following changes to my:
/etc/asterisk/extensions.conf:
[iaxtel700]
exten =>
_81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})
exten =>
_81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})
2010 Jun 10
1
Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?
Hi Guys,
I have Spikko setup as provider of DID and outbound routes and I can make
calls out but no inbound calls via DID can be made. I did a sip debug which
is reported below. I never receive the call though, I have a catch all in my
inbound routes and it doesn't hit my context at all or not sip invite comes
in:
FreePBX:
Trunk Name:
*Spikko*
Peer Detail
*username=MyUsername*
2014 Feb 08
4
force group does not work
Hi
I set up a samba 4.1.4 server on the latest FreeBSD RELEASE 10.
Unfortunately it doesn't seem to consider the option force group. After
hours ofresearch I couldn't figure out what I'm still missing. unix
extensions is set to no. Setting the debug level up to 10 also didn't
help ;(
Is this a bug or is there simply a mistake in my setup?
When
*valid users = @Groupname*
is
2007 Mar 19
1
"BadWindow" error w/ NV-GLX
Running wine 0.9.28, Ubuntu Edgy. nvidia 6800GT, dual-LCD w/ xinerama,
nv-glx driver.
Running winecfg always returns:
wine: creating configuration directory '/home/myusername/.wine'...
X Error of failed request: BadWindow (invalid Window parameter)
Major opcode of failed request: 144 (NV-GLX)
Minor opcode of failed request: 4 ()
Resource id in failed request: 0x24a
Serial
2018 Jul 28
2
Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
Using pjsip 2.7.2 on Asterisk 15.5
Really struggling to make sense of translating these old 1.8 SIP
instructions into a neat pjsip_wizard conf suitable for 2018
http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18
In pjsip_wizard.conf, I have the following, which seems to get me
registered, and it responds to an incoming call, but I always get
this:
[Jul 28 18:32:29]
2006 Feb 02
0
Sip - no peer or user found on incoming call
Hi list,
I try to connect to a GW which have one domain eg sip.mydomain.com and
have few IPs related to this domain. I register * to this domain with
host=sip.mydomain.com and type=user. So DNS will decide on which IP of
my domain I will register (or redirection on the GW side).
If an incoming call arrive, I would guess that, as type=user, it will
not try to match the IP from INVITE as I
2015 May 10
0
sssd on a DC
OK, I've got a little further and I think I have tracked this down to
a reverse DNS issue - which was non-obvious to me, so here is a
write-up for the benefit of the archives.
The part that was failing was this:
[sasl_bind_send] (0x0100): Executing sasl bind mech: gssapi, user: dc1$
[sasl_bind_send] (0x0020): ldap_sasl_bind failed (-2)[Local error]
[sasl_bind_send] (0x0080): Extended failure
2013 Jun 27
0
NTLM authentication mechanism with Postfix
I'm working on getting authentication for Postfix smtpd clients
working with Dovecot. I've got both plain text and GSSAPI mechanisms
working. Winbind also works for shell access and the command line
test work fine.
If I can get NTLM authentication working I can use Postfix as a drop
in replacement for a MS MTA I want get rid of.
I'm hoping the community might be able to offer some
2013 Jun 27
1
Dovecot NTLM Authentication
I'm working on getting authentication for Postfix smtpd clients
working with Dovecot. I've got both plain text and GSSAPI mechanisms
working. Winbind also works for shell access and the command line
test work fine.
If I can get NTLM authentication working I can use Postfix as a drop
in replacement for a MS MTA I want get rid of.
I'm hoping the community might be able to offer some
2013 Jun 24
0
NTLM Authentication for Postfix SMTP clients
I'm trying to get NTLM authentication working with Dovecot to
authenticate Postfix SMTP clients.
I can authenticate postfix smtp clients using the plain text login
mechanism through winbind. However, using the NTLM mechanism gives me
an error in my maillog that says:
"dovecot: auth: winbind(?,10.20.2.0): user not authenticated:
NT_STATUS_UNSUCCESSFUL".
At this point,
2013 Jun 25
0
NTLM Authentication with Dovecot and Postfix
I'm trying to get NTLM authentication working with Dovecot to
authenticate Postfix SMTP clients.
I can authenticate postfix smtp clients using the plain text login
mechanism via Dovecot and winbind. However, using the NTLM mechanism
gives me an error in my maillog that says:
"dovecot: auth: winbind(?,10.20.2.0): user not authenticated:
NT_STATUS_UNSUCCESSFUL".
At this
2005 Apr 05
1
Can't mount samba share, Access denied
Hello,
I have samba configured with the following smb.conf file:
[global]
workgroup = mydomain
netbios name = servername
security = domain
printcap name = cups
disable spoolss = yes
show add printer wizard = no
idmap uid = 15000-20000
idmap gid = 15000-20000
winbind use default domain = yes
use sendfile = yes
printing = cups
[myshare]
comment = My new share
path = /export/myshare
valid users =
2010 Oct 25
4
google voice + asterisk: calls made to GV# processed but weird
Dear all,
First off, I am very new to asterisk so forgive me if any of my
comments or questions seem trivial. Thanks to [this
post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/)
and [this post](http://www.davidvossel.com/?p=28), I have GV set up on
asterisk through jabber.conf and gtalk.conf. I can successfully dial
out from asterisk.
I'm trying to set up an
2002 Aug 27
5
rsync: push_dir TESTDIR: No such file or directory
Hi all. I'm getting the following error when using rsync:
nice -n 20 rsync -e "ssh -p30000" --recursive --verbose --verbose --checksum
--times --modify-window 2 --port=31000 --dry-run
/cygdrive/f/bkp/Doc/Builds/Buildsheets/ MYUSERNAME@MY.SERV.ER.IP:TESTDIR
opening connection using ssh -p30000 -l MYUSERNAME MY.SERV.ER.IP rsync
--server -vvntrc --modify-window=2 . TESTDIR