similar to: Most suitable version for Production ENV

Hi Once I installed astrisk , how do we connect with SIP interface ? Can somebody guide me how to integrate SIP interface with asterisk ? I want to use Astrisk just for IVR purpose. Thank you Luke -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130511/a6d32bfe/attachment.htm>

HI , I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files . What are the configurations has to be made with asterisk ? Thanx in advance, Luke. Send instant messages

Hi all ,? I have managed to install and configure the? 1. asterisk-1.8-current 2. dahdi-linux-complete-current I did not faced any issues during the installation. After that I installed X-Lite soft phone in two different PCs and tested the setup. every thing was success. I was able make calls from each?extensions. But when I observe the log files , i could see some messages ......

hi, please suggest me a mp3 source for icecast to run on ubuntu. it will be good if you can provide few options with decent installation and configuration. i tried darkice and its not working. With Best -Ashwin. +91-9959166266 -------------- next part -------------- An HTML attachment was scrubbed... URL:

When manually editing my files for production deployment do I need to edit .htaccess or routes.rb to include the images and stylesheets directories so they''re still found like in my local development environment? I''m FTP''ing my "appname" directory to my hosting service to deploy. Unfortunately they don''t allow ssh/capistrano access and have no

Does anyone know how to change timeout for the production environment? My action runs for ~2 minutes then timesout in production mode. In development mode it is ok. Thanks Chris -- Posted via http://www.ruby-forum.com/.

All, Just read in the Capistrano manual about the Capistrano "migrate" task. There it says that: "By default, all this task does is change to the directory of your current release (as indicated by the current symlink), and run rake RAILS_ENV=production migrate. You can specify that it should run against the latest release (regardless of what the current release is) by setting

Hello, I am testing for email sending from my app and I have a scenario that passes in test env but not in production. When I make dump of ActionMailer::Base.deliveries I saw that it returns array with duplicated elements in production env but not in test env. I am using sqlite3 for test env and mysql for production. Also I use delayed_job. Does anyone had the same issue? Thanks in advance :)

Hi? 1. When a mobile user dial an IVR short code , mobile network able to divert that call to Asterisk platform.? 2. There would be web servers which are holds Voice XML . 3. Asterisk would be able to redirect the mobile request to certain Voice XML server accordingly. Just for like this setup , how do we install asterisk with minimum of asterisk modules ? Do we need to install complete

I've been leaning towards actually deploying gluster in one of my projects for a while and finally a probable candidate project came up. However, researching into the specific use case, it seems that gluster isn't really suitable for load profiles that deal with lots of concurrent small files. e.g. http://www.techforce.com.br/news/linux_blog/glusterfs_tuning_small_files

Hi All, as I can see on https://github.com/defunkt/unicorn/blob/master/lib/unicorn.rb#L53 the?Rack::CommonLogger used only in development env and in weird evn called "deployment". Any chance to add "production" to this case? Serg Podtynnyi

hi, please suggest me a mp3 source for icecast to run on ubuntu. it will be good if you can provide few options with decent installation and configuration. i tried darkice and its not working. With Best -Ashwin. +91-9959166266 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of 2.0 due to a crashing issue resulting from ICE. https://issues.asterisk.org/jira/browse/ASTERISK-21696 Currently, I'm systematically going through each Makefile in every directory in pjproject and changing the paths that exist in the pjproject 2.0 included with Asterisk, so that I can successfully build

We have just updated our office server to Asterisk 11.4.0 from 1.8.15 and internally everything is working fine. The problem we are having is that we cannot use any external phone connected through the Internet. This used to work fine with 1.8 but since the upgrade whenever you register any phone from an outside network the phone tries to register using the servers internal IP. I endo up

Hi, I am using reticulate and a virtual environment (not conda) to run Python scripts from RStudio. However, when I try to use my own (existing) virtual environment, reticulate does not use it. If I run my scripts, the installed modules (e.g., py_install("pandas", "mmstat4.hu.data")) are not found. I believe this happens because reticulate is using r-reticulate instead of

Have you tried https://rstudio.github.io/reticulate/ ? Generally speaking, complex nonstandard package specific questions such as yours rarely get a reply here -- there are 20,000+ packages (and counting) after all! As reticulate was created by and integrated with RStudio/Posit, I would think their site and help resources might be a better venue. Of course, if you don't use RStudio, you may

I have been using 11.4.0 for some time. All was fine. I downloaded 11.5, extracted, run ./configure, make, make install. I got a message about res_rtp_asterisk.so was not compiled in the 11.5 Sure enough I have rss_rtp_asterisk.c but not .o file and no .so file. I then looked in the config.log and nothing is in there about res_rtp_asterisk What's up? jerry -------------- next part

cvs code from this morning (about 9am PST) breaks on slackware 7.1 w/ gcc 2.95.2.1 with an undefined reference to session_setup_sia in session.o. this seems to be the culprit here: #ifdef HAVE_OSF_SIA #else /* HAVE_OSF_SIA */ session_setup_sia(pw->pw_name, ttyname); since i have no idea what that's trying to accomplish (and seems to be a bit backwards to me from looking

First, I hope that this message is not considered flame bait. As someone who has used FreeBSD for for 5+ years now, I have a genuine interest in the integrity of our source code. Second, I hope that this message is not taken as any form of insult or finger pointing. Software without bugs does not exist, and I think we all know that. Acknowledging that point and working to mitigate the risks

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make