similar to: paginate with only some columns

Hi, I am using ajax_scaffold to generate against a model backed by atable on SQLServer... It creates all the stuff, and I can list the items in my table... But, the "Edit" and "Create New" buttons dont work... The strange thing is that the problem occurs when I edit the ajax_scaffold generated controller with before_filter/set_charset... I try to set the charset of pages

Hi, I am new to Rails. And I wonder how to define lookup fields for a table... Suppose we have an employee table and a departmant table... Each employee record have a department_id field... And when listing employees, we want to show the user friendly department_name from department table.. How can I do this? Thanks .... Bu e-posta mesaji, mesajin alici kisminda belirtilmis

Hi, I am using ajax scaffold generator with charset=iso-8859-9(Turkish)... Edit & Create buttons work correctly when there is no turkish chars in the current record. But, if there are turkish chars in any of the fields in the current row, buttons don''t work... (loading indicator simply spinning forever...) I put the KCODE=''u'' line in

hello, Our database is in is-8859-1, and I want to update some text fields without success due to some accentuate characters ?? ect ... In my html page (where the charset is iso-8859-19) my textarea display the accentuate characters well and when the user post the form ... I thought that I just need to save it .... without success since ruby map one byte for one character ... So I

Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli

You'll need one of the Digium Cards to make/receive pstn call...Chose the best one fits to you... To make calls over ip, voip phone is also needed such as grandstream, cisco, etc. or you can download x-lite from xten.com And use it with your computer as a soft phone... _____ From: infra struct [mailto:infrastructt@yahoo.com] Sent: Monday, June 06, 2005 1:01 PM To:

Hi; Can I use two TDM04B in one asterisk box with asterisk@home? Thank You Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi m?mk?n olmayabilir. Mesaji alan kisi, mesajin g?nderilmek istendigi kisi veya kurulus

Thank you and sorry...There is something going wrong with the system I only sent one mail... _____ From: Kerry Garrison [mailto:kerryg@techdatapros.com] Sent: Wednesday, May 11, 2005 5:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [SPAM] - RE: [Asterisk-Users] Grandstream-Budge tone - Email found in subject This is usualy a problem with either

I don't off-hand remember, but the folks on the user list probably do (;-)) You accidentally sent to the development list --dave Serhat G?RSOY wrote: > *********************** > This mail and its attachments have been scanned by InterScan MSS. > ***********-*********** > > > hi folks, > > I am a new samba user and I have a problem like this; > > When I copy

Hi all, I have been trying to implement speex echo cancellation algorithm on STM32F4 Discovery board; however, I could not make it work properly. I tried synchronous, asynchronous way of to do that and the only thing I heard was "cazzsdaos?eo?hodg etc" :) I try to use empty array for all parameters in the cancellation function, but the result is always the same

Hello again, I want to ask a question in my mind by reminding the topic below. Can the reason of failing be that Codec,which plays sound, and Mic are connected the different clock source? If it is because of that, what can I do to cancel the echo? Isn't there any solution for this? Is asynchronus speex functions for this? Best Regards, KAHYA PS: Sorry for my bad english. From: ?evket K?r?ad

Hi, I'm using Mandrake 10 CE, Samba 3.02a, Winbind. I have got a strange problem. My linux machines can not join to our NT 4.0 domain when I define a netbios name which includes "I" or "i". How can I solve this problem. Any answers .... Thanks a lot. Domain Name = DOMAIN1 Netbios Name = LINUX1

Hi, We have Ubuntu 14.04.3 server and Samba 4.1.6-Ubuntu. Windows and Mac Os X clients connect this server. Some Os X users sometimes report "error code -43" when file copying to server. What can cause the problem? this is my smb.conf [global] workgroup = hurguc realm = HURGUC.LOCAL preferred master = No security = ads password server = dbdns2.hurguc.local encrypt passwords =

I can't setup Authentication between samba & Win NT [root@fulya etc]# net join -U Administrator%pass [2003/08/12 20:22:55, 1] utils/net_ads.c:ads_startup(176) ads_connect: Connection refused [2003/08/12 20:22:56, 1] utils/net_rpc.c:run_rpc_command(154) rpc command function failed! (NT_STATUS_ACCESS_DENIED) Joined domain TRANSVARO. [root@fulya root]# What is ads_.. ? if it is

Hi all of you. I am trying to configure voice mail in asterisk and i am facing problems. I have found following warning message in /var/log/asterisk/messages -------------- No application 'Voicemail' for extension (macro-mainmenu, s, 5) I have configured voice mail accordingly in extention.conf [headoffice] -- ------------ ------------- exten => _63,1,Macro(mainmenu)

hello, i have trouble with nat + sip outgoing call.when make an outgoing call to a sip gateway, i have no sound. i have 2 sip gateway, one is asterisk. asterisk is on public ip and private ip other sip gateway is on public ip phone are cisco and grandstream on private ip on the same subnet as asterisk. phone are connected by sip to asterisk (i have try with or without nat=yes) incoming call

Dear useRs, I have a question with respect to fitting a non-linearity using gam (mgcv package, version 1.7-16). In a study I'm currently conducting, I'd like to find out if there is a breakpoint after which the effect of Age of Acquisition (AOA) of the second language changes. I.e. if the slope of AOA before the breakpoint (at a certain AOA) is different from the slope past the

I try to move from SPSS to R/S and am trying to reproduce the results of SPSS in R. I calculated a one-way anova with "spk" as experimental factor and erp as depended variable. The result of the Anova are the same concearning the mean square, F and p values. But I also wanted to caculate the contr.sdif(4) contrast on spk. The results are completely different now. I hope anybody can

Hi; Have two grandstream Budge tone...Connected them to the network and able to make call to/from them. But when the coming call answered, I can not hear any voice and also my voice is not heart... I am able to hear voice only if I pressed the hold button and take the call again....This problem also Occurs in calls from x-lite to cisco7940... Does anybody has any idea or documentation

> On 4 Dec 2017, at 11:58 , Levent TERLEMEZ via R-help <r-help at r-project.org> wrote: > > Dear Users, > > What is the proper way to write symbol, superscript, subscript in ggtern/ggplot? I tried every given example, every possible features of ggplot but couldn?t achived. I just want to write P_a, sigma^2, etc, would you please advise me about this problem. Did you try