similar to: Samba as PDC+OpenLDAP: unique login?

Hello people, My difficult is get "nss_ldap" compiled and running on my server. I download the tgz file from PADL Software Pty Ltd website, decompress/untar, and then run "./configure" . ./configure runs just fine with no error message. Things go bad when I type: make This is the output after I try to do "make": root@ldap:~/nss_ldap-262# make make all-am make[1]:

On 11.10.2016 17:22, Harry Jede via samba wrote: > Am Dienstag, 11. Oktober 2016 schrieben Sie: >> On 11.10.2016 13:52, Harry Jede via samba wrote: >>> On 10:43:49 wrote Gavrilov Aleksey via samba: >>> Until now, you have destroyed your domain. >>> Is the ldap directory on localhost in production or is this pc in a >>> test lab? >> a copy of the

Hello all=20 =20 First of all Sorry for the long e-mail =20 I am trying to get samba working as a domain member and store the idmap = in a ldap database. =20 The join is successful and all commands are working like it should = wbinfo =96u, wbinfo =96g kinit enz But the id administrator command gives me the following =20 # id administrator id: administrator: no such user =20 If I do

Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten => s,1,Dial(SIP/74118@193.136.252.5,30,r) but this way all calls go to 74118@193.136.252.5 ..... Then I tried: exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r) but this way, the

Hello, Realized update dovecot on my server. Now the search is returning differently from the previous version bringing reference information of other messages . For example when doing a search for anderson.joao this new version of the dovecot dovecot 2.2.21 + fts - Solr response will be all email that has the word anderson and joao, instead of returning only items with the word anderson.joao.

Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten => _555,1,ChanSpy(Agent) exten => _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which

I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close

Hi, I would like to know how can I keep adding data to a file while doing a loop and without deleting the data of the previous iteration. Thanks.

Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao

Hello to all Im looking for a billing tool for Asterisk, that works with PostgreSQL. All the tools I found in www.asteriskbilling.com just work with MySQL :( Do you know a nice billing tool for Asterisk with PostgreSQL? Thanks Joao Pereira

Hi all, I was wondering if someone can give me some pointers to a good Document Management System. I've been using Knowledge Tree for a while and just wondering if there is something a bit more robust. I'm archiving technical specs, functional documents, etc... TIA, Joao Jo?o Medeiros Linux User 381318

Hello I'm trying to use an AGI that returns the queues status (numbers of available agents, etc ), but I'm having some problems with it (it's still very buggy). Is there any AGI repository with source code samples? Had anyone used an AGI to check queues and agents status? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip:

Hello I'm using eyeBeam, and Asterisk keeps sending my clients a second call, when they are still in one call (because eyeBeam has lots of channels). I was using X-Lite (with 3 channels) and Asterisk never sent the client a second call. How can I force Asterisk (or eyeBeam) just to send one call at each time. Is this a configuration I need to do in eyeBeam or Asterisk? Thanks Regards Joao

Dear all, I would like to know how can I create a binned column in a data.frame. The output that I would like is something like this: Start Binned_Start 1 0-5 2 0-5 6 5-10 8 5-10 13 10-15 ... Best regards João Fadista Ph.d. student UNIVERSITY OF AARHUS Faculty of Agricultural Sciences Dept. of Genetics and Biotechnology Blichers

Dear all, I would like to know how can I retrieve a data.frame without the columns that have a partial match name. Let´s say that I have a data.frame with 200 columns and 100 of them have the name "StartX", with X being the unique part for each column name. I want to delete all columns that have the name starting with "Start". I´ve tried to do this but it doesn´t work: >

Hello I'm trying to register an Asterisk working behind Nat. Here is the trunk: register=username:password at sip.startel.pt [startel] type=peer host=sip.startel.pt username=username fromuser=username secret=password qualify=yes disallow=all allow=ulaw allow=alaw allow=gsm insecure=very port=5060 nat=yes canreinvite=yes The problem is: Asterisk is registering with its

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

Here are my scopes: default_scope :order => ''posted_on DESC'', :conditions => { :status => ''visible'' } named_scope :positive, :conditions => { :rating => ''positive'', :status => ''visible'' } named_scope :neutral, :conditions => { :rating => ''neutral'', :status =>

I''m starting now with rails, and i have simply question i think. I need to render two partials in one ajax call: I have the following controller: # GET /hosts/1 # GET /hosts/1.json def show @host = Host.find(params[:id]) respond_to do |format| format.html #show.html format.js format.json { render :json => @host } end