search for: schmoes

Hi! I'm struggling with pam_mkhomedir. Accounts are stored in LDAP (CentOS Directory Server AKA 389 Directory Server). I am not pre-creating user home directories, and would like to use the pam_mkhomedir module to create the directories on the fly, when a user is authenticated via pop3s. pam_mkhomedir is working for interactive (e.g. SSH logins), but I don't want to require users to

For those of you interested in getting a jump on learning about and using the new Story Runner in a Rails app, I''ve added an experimental Rails-Story adapter to trunk. You can see how to use it here: http://pastie.caboo.se/92320 This is PURELY EXPERIMENTAL, however it actually works and will likely not change much in the way that it does work. Happy hacking! David

Good morning (Pacific Time). I have a controller action that, as a side-effect, sends an email to an administrator. I want it to do something like this: specify "when someone successfully signs up, an email should be sent to the administrator with the person''s contact page" do post :signup, {...lots o'' params} response should_be success #

I have a spec: specify "a post with no user id should add a record" do post :edit_or_create, :user => {:login => ''joeschmoe'', :email => ''joe at shmoe.com'', :email_confirmation => ''joe at shmoe.com'', :full_name => ''Joe Schmoe''} assigns[:user].should_not_be_nil

Hello, I wish to use asterisk as a voicemail server with ser . I want to use asterisk external configuration toHello, I wish to use asterisk as a voicemail server with ser . I want to use asterisk external configuration to manage users and storing voicemail messages according to ser database. Where can i find the schema of the SQL DB for voicemail accounts . for example in extconfig ;

Folks, I'm experimenting with bringing multiple (8) analog lines from our local telco into a Carrier Access ADIT 600 channel bank with an FXO module, then having this talk to Asterisk via the T1 TDM controller on the ADIT and a TE405P card. I don't know if this will work well (ie: give me decent echo cancellation, call disconnect supervision, caller ID etc) but I haven't had much

Had an appointment for these schmoes to come out and install another line. Was supposed to be 8-12. Its now 6PM and not even call. Missed 3 sales calls waiting on these jerks. No wonder customers were jumping ship to Vonage. -- Warm Regards, Lee

Hello Samba People, We have a Kerio Connect (email) server using Samba 4.8.5 as it’s directory service (3 AD DCs). We’ve been using this setup for about 3 years now. Over the last several months, we’ve been trying to find out why Samba starts rejecting attempts that the Kerio Connect mail server makes to authenticate our users. The errors in Kerio look like this: Authentication failed for user

Folks, I recently swapped a TDM400 FXS card that was working perfectly into a new server (running recent CVS), and it's either misbehaving (unlikely), or I've missed something obvious (much more probable). Everything seems to be working, but I can't get any dialtone from it when I plug a phone into any of the 4 ports!! All of the jacks are lit (green lights) but they all seem dead

Sending this to asterisk-users, in case anyone has AsteriskNOW experience they can share. Joe ---------- Forwarded message ---------- From: Joe Wood <schmoe at gmail.com> Date: Wed, Aug 18, 2010 at 9:22 AM Subject: AsteriskNow REGISTER'ing s@ extension for all inbound trunks To: asterisknow at lists.digium.com Hello. Can someone tell me why AsteriskNow is reverting to registering

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

Thought a different succinct subject line must drum up an answer or two... Also, this has been tested from two different carriers: We're getting an average of 2/10 call success rate. ---------- Forwarded message ---------- From: Joe Wood <schmoe at gmail.com> Date: Thu, Aug 26, 2010 at 6:58 PM Subject: Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround()

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, is it really necessary to add every user to the administrators group to allow them to update their windows boxes? I really can't believe this... Cheers Nicki - -- Linksystem Muenchen GmbH info@link-m.de Schloerstrasse 10 http://www.link-m.de 80634 Muenchen

Hello, I built apache+openssl+mod_ssl. It is working fine, and I have been starting the server with: apachectl startssl Recently, however, I have decided that I will not be doing anything over https (for a while, at least) with this web server, so for security reasons, I want to only run on port 80. So now I start the server with: apachectl start And it runs without SSL. My question is, is

Hello All, I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy static. Even with the pots line disconnected, if I do a dial I still get static. This way I know it's not the line, but rather something on the card. I tried alternate pci slots. This card has a power connector, does anyone know what the power requirements are? The unit is in a small case with a 2.4ghz p-4 and

I have gotten acts_as_taggable to work for a test application as documented on taggable.rubyforge.org I am following Chad''s excellent examples in ROR Recipes Beta book. I have a few questions and hope that Chad or some other expert can help clarify them. 1. Why is the book suggesting to use taggings table instead of tags_contacts, as mentioned on taggable.rubyforge.org?

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details