search for: bartend

I am trying to get Rails running through Lighttpd, on a Suse 10 box running Rails 1.0.0 and Lighttpd 1.4.10 I followed the instructions in the wiki (http://wiki.rubyonrails.com/rails/pages/Lighttpd) but keep getting the same error: linux:/etc/lighttpd # lighttpd -f lighttpd.conf 2006-02-27 12:32:17: (mod_fastcgi.c.997) execve failed for:

...ne in the dialplan who is the transferer without having each phone in it's own context, that would be fine. -- Automated Signature: This message is from Flobi of Flobi.com<http://Flobi.com> . Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050907/8da7a475/attachment.htm

I am communicating with Asterisk 13.18.3 over the AMI and issue the command: ActionID: 11 Action: command Command: core show calls And the response I get is: Response: Follows Privilege: Command ActionID: 11 --END COMMAND- But where is the call data? What is going wrong on this system? I confirmed the AMI connection has full read/write permissions. Why is the call data

Hi, I''m trying to write my first custom matcher. Here''s a bit of my example group. describe "/contact/index" do include FormMatchers before(:each) do render ''contact/index'' end it "should show the contact form" do response.should have_a_contact_form end describe "the contact form" do context

...; Recently, someone who shall remain nameless tried names(mydata) <- "d^2" followed by > mydata$d^2. As variables in a data frame are generally considered similar to variables > in, say, the global environment, it seems strange that foo$bar can give you the content > of foo$bartender. In R-devel (i.e., *not* R-3.0.0 beta, but 3.1.0-to-be) partial matches > now gives a warning. Of course, it is inevitable that lazy programmers will have been > using code like >> > [[alternative HTML version deleted]]

Hi, I'm using the Record dialplan Application in an Context. My goal is to get a single screenshot of the h264 media stream per call. same => n,Record(/tmp/test.wav,0,10,qk) I nicely get a File test.h264. Is there a way to Playback this h264 video file on my computer or convert it somehow? VLC can't take it somehow. Regards, Benjamin -------------- next part -------------- An HTML

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Wed, 19 Mar 2014, Steffen Kaiser wrote: > IMHO, the behaviour matches your config. If my assumption in my previous message is correct, you will have some options: a) have UserDB return "mail", b) make mail_location depend on home via ~ c) create a symlink default location -> public d) forward office to some other user where you

...aDSL connections running. And both run the Asterisk PBX software, I defined the IAX trunks in the iax.conf on both sides, the systems seem to call eachother but never get the handshaking completed ... The consoles keep reporting errors on both sides. Is there a example for doing this ?? -- A bartender is just a pharmacist with a limited inventory. I used to have a handle on life, but it broke --- Robert Bungener Dutch World Ventures

Address lookup canonical name asterisk.org. aliases addresses 216.27.40.102 Service scan FTP - 21 Error: TimedOut SMTP - 25 Error: ConnectionRefused HTTP - 80 Error: ConnectionRefused POP3 - 110 Error: TimedOut NNTP - 119 Error: TimedOut digium.com is ok though Address lookup canonical name digium.com. aliases addresses 216.207.245.1 Service scan FTP - 21 Error: TimedOut SMTP - 25

Dear all, Has anyone seen this before and can suggest a solution? I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. Could

Hello, I'm still looking for any ideas on this problem: I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. I have

I apologize for the double post. I am curious as to what the usefullness is of the multiple line appearance feature on Polycom phones. I setup our phones to register one line per extension but I hear the IP501's can do three line appearances. Why and how could this feature be applied? Thanks again all. Kenny ______________________________________________________ Click here to

How do I set each extension to play it's own voicemail prompts? I have vm working in that it plays the standard "person at extension 1234 is not available....." and takes the message. I've recorded seperate .gsm files for each user but can not figure out how to use them. - Gary Edison Information Technologies www.EdisonInfo.com P.O. Box 554

Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775)

http://vuc.me Kamailio, Open SER and Asterisk walk into a bar... The bartender is Alex Balashov, someone whose posts I have long admired on this list. Alex has agreed to take us through the following areas: - Relationship of Kamailio to OpenSER project history. - What is Kamailio/OpenSER? - SIP proxy - SIP server (for certain purposes, such as registrar, presence user...

...predictions, especially about the future"-- Yogi Berra. On 03/21/2013 06:00 AM, r-devel-request@r-project.org wrote: > As variables in a data frame are generally considered similar to variables in, say, the global environment, it seems strange that foo$bar can give you the content of foo$bartender. > > In R-devel (i.e.,*not* R-3.0.0 beta, but 3.1.0-to-be) partial matches now gives a warning. > > Of course, it is inevitable that lazy programmers will have been using code like [[alternative HTML version deleted]]

I''m in the process of documenting templates right now, and I figured I should see what happens when you use them with arrays: $ cat ~/bin/test.pp $values = [this, is, an, array, of, values] $content = template("/tmp/templates/testing.erb") file { "/tmp/temtest": content => $content } $ cat /tmp/templates/testing.erb <% values.each do |val| %> I got

Hi, Is it possible to have an array as the output of a custom fact? And then to pass it into a template in Puppet? I currently have a fact that looks like this: Facter.add("exports") do setcode do case Facter.hostname when (/thishost/i): [ "/local", "/local2" ] end end end & a template like this: <%

Hi all! I would like to solve some problems: I have a sip provider that lets me make pstn calls after listening some stuff and entering a pin number: 1) How can I make Asterisk enter the pin number? Then wait 1 second and enter the phone number? I have in extensions.conf: exten => 6*,1,Dial,SIP/2002@myprovider,60,tr I have tried with w (like with ZAP channels) but it does not work, nor

Hi, I'm working with this issue for a while, Now I already solve the dialplan issues, but I still have a question about the Callgroups, I read at www.voip-info.org that , there is a 63 limit of callgroups. And I'm wondering why?? and if the 1.2.0beta version supported more than 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any unoficial patch for that ? Thanks