similar to: Tests in VMWare

Forgive me if this has been covered before. I did search but I was unable to find a reference. I am curious to know more about the possibility of using SQL to store voicemail as well as having more than one voicemail system accessing a central SQL database. Any information would be appreciated. Thank you all, in advance. -- Ein Bielaczyc <ebielaczyc at gmail.com> NOTICE: This E-mail

I just loaded a new asterisk install (1.4.19) and found that the sip, iax, and extentions commands are missing from the cli and are not listed in help either? Any idea what could have happened or where these commands may have gone? Thanks, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Mon, 2008-03-31 at 03:04 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Mon, 31 Mar 2008 07:55:08 +0300 > From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> > Subject: Re: [asterisk-users] Tests in VMWare > To: asterisk-users at lists.digium.com > Message-ID: <20080331045508.GT12487 at xorcom.com> > Content-Type: text/plain; charset=us-ascii >

Hello, I was tasked to install Asterisk 13.16.0. from source on a CentOS7 platform. For that purpose, I used an unmaintened script of mine, written 10 monthes ago, and I was surprised to get segmentation violations whenever I ran "asterisk -cvvvvvvv -U asterisk". Usually, my /etc/asterisk/modules.conf file includes "autoload=yes" setting. Basically, I see two alternative

Hi all, I have a demo machine I'm running up on Lenny - it has the packaged Asterisk version installed (1.4.21.2+stuff). I'm trying to add an extension to leave a voicemail message, just with Voicemail(1234), which I've done before (on 1.2 at least), but it's saying "no application 'Voicemail' ". "module show like voi" shows

I am having a problem with a couple servers. They both have a Digium TE110P and a TDM04B card. I have setup the system so the TE110P uses channels 1-31 and the TDM04B 32-35. The problem is that when we reboot the server sometimes the TDM04B is recognized first and the TE110P second so the configuration fails. I do not know why this happens and to solve this I have to do a "service

Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with

hello while i try to compile zaptel it gives following error to me you do not appear to have the sources for the 2.6.27-7-server kernel installed can anybody know i have vmware and using centos 5 regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090912/54ce441e/attachment.htm

Hello, I have started using Asterisk recently on my Ubuntu server. I installed it first using apt-get and it worked fine sort of, but still couldn't hear voice during the call! I read that this problem solved by reinstalling it, so I decided to reinstall the latest version from the source as apt-get can give you only Asterisk 1.8So I have installed Asterisk 11 following the procedure in

I used the command asterisk -vvvvvc to see console messages and it works fine. Now, I want to turn off this feature. How please?

Hi list, Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run into some serious problems. The first thing I noticed was this message that would show up every five seconds on the CLI: Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! == Primary D-Channel on span 1 down

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

Hi Luke and other people working on 0.24. I hope this is constructive. So, I have our configuration working against a checkout from git as of last night. Subject to (sorry, here''s the big but :) * external_nodes script is not being called (#951) * Custom types aren''t working unless I drop them into $rubysitedir/puppet/{type,provider}. Previously they worked in

Hi All, I recently installed and compiled Asterisk in a Red Had (enterprise) box with digium TDM400P card installed in it. Thanks to the good documentation available, I was able to install and configure the hardware as well as software. As specified in the Getting Started with Asterisk Guide, I first installed zaptel driver, then libpri and finally Asterisk. All of them successfully compiled, but

I can't get Emacs to automatically do syntax highlighting of Sweave files. I have followed Friedrich's suggestion for code to insert into my .emacs file. The complete section from my .emacs file is given below. When I load a *.Snw file, font is white until I press M-x, then the first code and document chunks get highlighted, but not the rest of the file. Latex and Noweb menus are

Does anyone know whether the linux machine running * have to have a sound card on it in order for musiconhold to work for sip phones? I've tried about everything (including tons of google searching) to get it to work, and nothing. When a call is placed on hold between two C7960's, the CLI indicates: -- Executing Dial("SIP/3002-c418", "SIP/3000|20") in new stack

Hi Guys, I''m unable to load the module cloud provisioner, when I try to run node_aws it gives me the below errors: Error: Could not autoload puppet/face/node_aws/list_keynames: no such file to load -- guid Error: Could not parse application options: Could not autoload puppet/face/node_aws/list_keynames: no such file to load -- guid And when I see the help list I can see the below

I've built the latest CVS of asterisk -- not the zaptel or libpri directories, just the asterisk directory. asterisk installs successfully, but there are severe problems. I built this system in the past and ran it, but now building it again fails. This is the CVS as of this morning, 2003-06-13, but I had problems on 06-11/12 as well. After make; make install; make samples; make config, I

Good day, After having lurked for a while I''ve decided to actually start using puppet starting on my small home network. I''ve been following : http://reductivelabs.com/trac/puppet/wiki/InstallationGuide and :- http://www.howtoforge.com/installing_puppet_on_ubuntu to test puppet on 2 hosts, librarian running the puppetmasterd and mortvm. Unfortunately, I

Trunk Issue: Because of the use of ''autoload'', template handlers other than Erubis are not loaded automatically (Haml, XMLBuilder). Either this should be fixed, or the documentation should be updated to instruct people how to use non-Erb template engines. Apparently the solution is to do something like this in merb_init.rb: ::Merb::AbstractController.register_engine