similar to: Change time causes profile corruption

I have Samba 3.0.9 running on SuSE 9.2, 2.6.5-7.111-smp kernel. SAMBA is a PDC using OpenLDAP as a passdb backend.??Workstations?are?combination?w2k SP3, SP4, and Windows XP SP1. The problem I have is with profile synchronization. If?a?user?obtains?a?file that has a blank modified time, Windows substitutes Jan 13, 2038 for the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 After a failed migrate from NT4 to Samba 3.0 using the rpc vampire method I decided I'd do it manually with 'net rpc samdump'. Well it sort of worked, except I can't edit any of the users settings, it might just be me, but it should modify the profile to that, just as a test but it doesn't. I'd actually like to get rid of the

Hi. We have an odd problem with incoming SIP calls. I have attached a SIP debug log, with some asterisk verbosity as well, demonstrating the problem, below. Is this a known bug? Vital stats: - Asterisk 1.2.3 - Sipura SPA-841, SPA-941 phones - Fedora core 3 The problem manifests itself with these symptoms: - an internal SIP extension receives a call from our PRI - the SIP phone answers the

Hello: I'm using: - samba-common-3.0.9-1 - kernel 2.6.5-1.358 - FC 2 - openldap-servers-2.1.29-1 We're running an NT4 domain using an LDAP backend, and everything was running fine until recently. The first thing that I noticed that new users were suddenly being assigned SambaSID's that were previously being assigned to machines. Previously: Typical User Entry: uid: john

Hi, I am running a couple of Samba / LDAP servers. While they all do work fine, I get a message like this on all of them when I run pdbedit -L -v: Unix username: administrator NT username: administrator Account Flags: [UX ] User SID: S-1-5-21-XXXXXXXXXX-XXXXXXXXXX-XXXXXXXXXX-21000 init_group_from_ldap: Entry found for group: 512 lookup_global_sam_rid:

Return-path: <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx> Envelope-to: xxxxx at xxxxxxxxx Delivery-date: xxx, xx xxx xxxx xx:xx:xx +xxxx Received: xxxx [xxx.x.x.x] (xxxx=xxxxxxxxx) xx xxxxxxxxx.xxxxxxxxxxxx.xx xxxx xxxxx (xxxx x.xx) (xxxxxxxx-xxxx <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx>) xx xxxxxx-xxxxxx-xx xxx xxxxx

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

I have a data set with observations on 549 cities spanning an 18 year period. However, some of cities did not report in one or more of the 18 years. I would like to implement the procedure suggested by Wooldridge section 17.1.3 in his "Econometric analysis of cross section and panel data" to correct for attrition. For example the table below indicates that the 3rd and the 7th cities in

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it. The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail.

Hello, I have a SLES10 64bit machine running samba 3.5.12. i have configured a domain (TASC), and set the servers netbios name to TUX. Samba is using the tdbsam backend. Now I have add machines to the domain, and users can logon using their domain accounts TASC\user. However, if the network is disconnected, in the case of laptop users going offsite, they cannot logon anymore. I have now

I have a root cron job that powers down my server every day at 1am and 6pm. The output of '# crontab -l' is shown below. * 1,18 * * * poweroff Last night, after the server powered down at 6pm, I decided I wanted to use the server so I started it with the power button. The server, after a minute or so, powered itself down. This behaviour happened repeatedly until I waited past 7pm.

First off this is not my work for extensions.conf it is modified from http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten =>

Hi all, a noob here, I am trying to get outbound calls through asterisk working with Broadvoice. I have consulted the following two online tutorials: http://www.broadvoice.com/support_install_asterisk.html http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice in an effort to make outbound calls. My current settings are as follows: sip.conf register =>

I have downloaded and installed Asterisk@home and I have installed X-Lite on my Windows machine and I am able to connect it to the Asterisk server. I went ahead an created an account on Broadvoice today and followed the directions on http://voip-info.org/wiki-Asterisk+settings+Broadvoice and http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but when ever I try and make a call from

Hi, I've setup samba 3.4.7 to use idmap_rid as per the documentation: idmap backend = rid:DOMAIN=500-100000000 idmap gid = 500-100000000 imap uid = 500-100000000 It seems to work for groups: wbinfo --group-info="domain admins" domain admins:x:100512 PsGetSid v1.43 - Translates SIDs to names and vice versa Copyright (C) 1999-2006 Mark Russinovich Sysinternals -

Hi Steve, - Your right - the file was AMI (copied the other one). By direct connect I simply meant the speaker is an extension on that server. here is the SIP debug <--- SIP read from UDP:X.X.X.X:1024 ---> == Using SIP RTP CoS mark 5 Audio is at 16060 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably

Hello list- As a FreeBSD shop we've used Samba 3.x quite well for a couple years. With version 3.6 due to expire in due time, we've been experimenting with version 4.1 using winbindd with very limited success. We find that if we use the TDB backend instead of either RID or AD, we are able to enumerate our AD users via getent. I cannot enumerate AD users via either the AD or the RID

On Mon, 25 Feb 2019 11:19:33 +0100 Viktor Trojanovic via samba <samba at lists.samba.org> wrote: > > On 25.02.2019 10:20, Rowland Penny via samba wrote: > > On Mon, 25 Feb 2019 09:24:24 +0100 > > Viktor Trojanovic via samba <samba at lists.samba.org> wrote: > > > > > > > >>>> I'm confused.. how is the choice of the idmap backend

It gets hard to read my logs when every time someone makes a phone call it displays long pages of "Dropping voice frame". Anyone encounter this before? Asterisk is bridging two SIP lines together, so the technology should be the same. Maybe I'll try allowing only ULAW. ************************************** Asterisk Standard debug (level 3)