similar to: Ubuntu 12.10 Nginx Rails 3.2.13. Deploy in sub URI. Nothing happens!

I''ve written a simple Rails migration that adds an index to an existing Postgres table: class AddIndexToEvents < ActiveRecord::Migration > def change > add_index :sf_events, :account_id > end > end However, when I run the migration, it fails due to a syntax error on the CREATE INDEX line. I''m expecting some SQL like this to be generated: CREATE INDEX

Hi there - I''m a ruby newbie and am stuck on the Getting Started with Rails app. See output below. Problem arose when I ran the rake db:migrate command. I also tried the ''rake db:migrate RAILS_ENV=development'' command which may be obvious. Any ideas on how to fix this? Many thanks!! $ ruby -v ruby 1.9.3p392 (2013-02-22 revision 39386) [x86_64-darwin10.8.0]

I have RoR running on lighttpd on RedHat Linux and every morning that I check the web page it just keeps waiting to refresh it forever, Firefox and IE never times out on any of the pages I try. Has anyone seen that happening to their RoR? I also removed lighttpd completely and just ran webrick but there was no difference. The linux box has a pretty old kernel 2.4.21-4, could that be the

I am struggling to understand the methodology here... I have a find.rhtml <%= start_form_tag :action => ''list2'', :first_name = client %> <%# render :partial => ''form'' %> <p><label for="client_first_name">First name</label><br/> <%= text_field ''client'', ''first_name''

Dear All, I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk??

I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show

I just noticed the Aastra 57i do something that I haven't seen before. I called from one phone (phone 1) to the 57i. I answered it. Then, I pressed Transfer and dialed the extension for the third phone (in this case a Cisco 7960 in Sip). I did not answer the Cisco, but noticed the caller ID was showing the Aastra (as expected). I hung up the Aastra to complete the transfer and noticed the

I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. I've looked at the sample config files for 1.4 and nothing seems to jump out at me as to what

éÚÍÅÎÅÎÉÑ × nginx 0.8.47 28.07.2010 *) éÓÐÒÁ×ÌÅÎÉÅ: ÐÅÒÅÍÅÎÎÁÑ $request_time ÉÍÅÌÁ ÎÅ×ÅÒÎÙÅ ÚÎÁÞÅÎÉÑ ÄÌÑ ÐÏÄÚÁÐÒÏÓÏ×. *) éÓÐÒÁ×ÌÅÎÉÅ: ÏÛÉÂËÉ, ÐÅÒÅÈ×ÁÞÅÎÎÙÅ error_page, ÎÅ ËÜÛÉÒÏ×ÁÌÉÓØ. *) éÓÐÒÁ×ÌÅÎÉÅ: ÅÓÌÉ ÉÓÐÏÌØÚÏ×ÁÌÓÑ ÐÁÒÁÍÅÔÒ max_size, ÔÏ cache manager ÍÏÇ ÚÁÃÉËÌÉÔØÓÑ. ÏÛÉÂËÁ ÐÏÑ×ÉÌÁÓØ × 0.8.46. -- éÇÏÒØ óÙÓÏÅ×

Изменения в nginx 1.23.4 28.03.2023 *) Изменение: теперь протокол TLSv1.3 разрешён по умолчанию. *) Изменение: теперь nginx выдаёт предупреждение при переопределении параметров listen-сокета, задающих используемые протоколы. *) Изменение: теперь, если клиент использует pipelining, nginx закрывает соединения с ожиданием

I am looking to replace all of the old "Bell" (POTS) phones in my home and office with IP phones. As you can imagine I don't have a huge budget to work with but I want phones that will provide acceptable voice quality and durability. There are basically three categories as I see it 1. satellite phones (low cost, low function) 2. primary domestic phone (good quality, POE capable,

I have a patch that enables "user password timeout" for 2.2.2. It works only for WinNT, Win2000 (tested and Samba on AIX and Linux) and I think with WinXP. It does not work with Win9x. You can download the patch from http://www.m-wohlfarth.de/Downloads/Samba/samba.html. I never had the problem with automatically expiring password as described in the originating message - I couldn't

On the server [root@bangvmpllDA02 logs]# ruby -v ruby 1.8.7 (2011-06-30 patchlevel 352) [x86_64-linux] [root@bangvmpllDA02 logs]# puppet --version 3.0.1 and [root@bangvmpllDA02 logs]# service nginx configtest nginx: the configuration file /apps/nginx/nginx.conf syntax is ok nginx: configuration file /apps/nginx/nginx.conf test is successful [root@bangvmpllDA02 logs]# service nginx status

Hello, In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system. Is there a way to manipulate this message, as well? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

Lacy's response in the thread 'Why does everyone seem to dislike *now?', has a small bit that caught my eye. Chan_Skinny made a lot of progress between 1.2 and 1.4, and even more in the later 1.4.X releases. I am curious as to which features/functions that chan_skinny might be lacking compared to chan_sccp. We (the community) now have a small, but active, group of volunteers

The "telco guys" probably did something non-industry standard and reversed send and receive in the jack that they plugged the CAT5 into. Sure it works, sure it is easier, sure it is not the correct way of doing things. Thanks, Steve ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Lacy Moore - Aspendora Sent: Sat 4/22/2006 2:55 PM To: Paul

I would like to get in contact with users/consultants who are or have worked with the Cisco phones and Asterisk to trade information. Cisco has reluctantly made SIP available on their phones and most of the information on voip-info and other wiki's appears to be reverse engineered. There is a wealth of information out there which is terrific. I have a client with about 40 phones

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

Hi: Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox server isn't seeing the mainboard's APIC. -Stephen-