similar to: 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

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,

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

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, I'm

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

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

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.

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

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

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})

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*

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

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

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]

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

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

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 =

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

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

We recently upgraded to rspec2 / rails 3 and our specs are taking a very long time to run. It takes nearly 12 minutes before the first dot comes out. Before the upgrade, our entire suite of 6000+ specs would run in about 6 minutes. The majority of the delay seems to be during the loading of the spec files. Adding instrumentation to the rspec-core lib/rspec/core/ command_line.rb, you can see