search for: find_users

Hi All, This is the first time I'm submitting a patch, and I hope it fixes more than it breaks. I'm putting it here, since John Todd mentioned a while ago about the heavy load Mark and crew have at Digium (doing such good work), so I thought all of us could test this first, and if ok submit for inclusion in CVS later if appropriate. This is an extension to work done earlier (sorry I

I''d like to prompt a user for the value of a Name field, then display the record. Rails tells me that it cannot do a find without an ID. I guess it must be that I''m not passing back properly the data from the view to the controller. Thanks for the help joshi The find_user.rhtml view: <div class="find-name-form"> <fieldset> <legend>Enter User

I have some code working that lists only items from a particular user. The code in my list action finds the user and then conditionally lists only his/her items: def list user = User.find(session[:user]) user_id = user.id @product_pages, @products = paginate :products, :per_page => 10, :conditions =>[''user_id = ?'', user.id]

on the main layout for my controller, i need a list of users - each with their own link to a lists of tasks that they have been assigned. i did have a line inside the .rhtml file that found all the users, but i''m guessing this is probably the wrong place to put them. i''m new to ruby and rails, so i still struggle a little with the MVC concept. -- Posted via

I forgot to take out the portion that would read in the voicemail boxes from the text file. If you want to leave it in then you could have some voicemail boxes defined in the text voicemail.conf. I do not, so I have removed it. Below is the new patch: *** app_voicemail.c 2004-06-23 07:55:54.000000000 -0600 --- app_voicemail.c.new 2004-06-23 07:55:47.000000000 -0600 *************** *** 49,61 ****

Hello all, I am just getting going on building my system, but I thought I'd send you all a patch that I wrote so the command: show voicemail users issued from the CLI works properly when there is a postgres backend for the voicemail. The current version of the app does not display the voicemail boxes found in a database. It is called in the load_config function. I haven't done

Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

I have what I hope is a simple question regarding a security practice I''ve been using in my first Rails app. I want to know if it''s worthwhile or if the extra typing isn''t worth it. I have 3 models that are related to each other. class User < AR:Base has_one :library end class Library < AR:Base belongs_to :user has_many :items end class Item < AR:Base

...t I am targeting, the plain html elements from the partial template have been stripped, leaving only the elements inserted via the <%= statements. I would appreciate any hints. Best, Eric here is the controller action that responds to the ajax request: =============================== def find_users @user_is_admin = is_admin @user_pages, @users = paginate :users, :order_by => ''login'', :per_page => 10, :conditions => "name like ''%kyle%'' or children like ''%kyle%''" # add error checking later ren...

Hi all. Trying to setup H.323 via Asterisk between a PLANET H.323 box and my SIP phones. When calling from the SIP phones, it connects but quickly disconnects citing the following error message: **** --- build_peer +++ build_peer +++ reload_config +++ ooh323_do_reload -- Executing Dial("SIP/yyy-2965", "OOH323/203@xxx") in new stack --- ooh323_request - data

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

Hi, Some calls I make trough my PSTN asterisk gateway just hangup after some minutes. Even if I'm using sip or iax. I have callprogress=no busydetect=no in my zapata.conf. Anyone help? Or tell me what to look at /var/log/asterisk/debug. I didn't find anything wrong. [endpoint]---iax or sip----[asterisk]----E&M----PSTN. As endpoint I had tested another asterisk box (with a FXS),

I've been looking at how our budgetone's have been failing and have found the following: A quick layout -- Latest CVS as of tonight. Sip phone behind NAT. * server with public IP address. -------from sip.conf for my phone: [1747xxxxxxx] username=xxxxx secret=xxxxx host=dynamic type=friend nat=yes ------- -------from the * log messages Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c,

Hi all, I have the following setup: UAs ------------SER ------------------------ ASTERISK ---------------------GNUGK --------------- GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a

Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message "Didn't get a frame from channel: SIP/3805-df43", but I can't figure why. asterisk logs:

I happened to mix ryan bates'' authentication scaffold with rspec_scaffold on a demo project. and ran into the problem of mixing mock frameworks...ryan uses mocha. So, as a learning experience, I choose to redo ryan''s tests without mocha but ran into a strange problem with tests of the User model. With debugging you can see.... If you run just the user_spec.rb file, everything

Hi, I am investigating an issue with voicemail and realtime. What we are seeing is the following: 1. Caller calls in and goes to an IVR 2. Presses 101 to go to voicemail 3. app_voicemail start and tries to connect to the database trhough res_config_mysql. However, it takes too long to be able to connect (~15 minutes) It seems like it first attemots to connect to the database on 16:25:03 and

(I have some problems with my mailing-list alias, I hope this doesn't get sent twice) On Sat, Nov 15, 2003 at 04:35:20PM +0100, Philipp von Klitzing wrote: Thank you for your comments Philipp: > > - with a SIP phone configured as 192.168.1.190, and with its SIP > > server being 192.168.1.190 > > That doesn't look right. Do you have another "SIP