similar to: question about more than one drop file

Dear list, I am trying to configure the nagios plugin called check_sip. I just read the README file included with the plugin. I follow the readme steps to configure the plugin, without success. In the nagios web interface I can see (No output!) In the status information column. If I run the chech_sip plugin from a linux console, I get /usr/local/nagios/libexec# ./check_sip -u

Hi, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated.

hi, I'm getting registration errors I can't debug... [Mar 23 11:07:20] NOTICE[2952]: chan_iax2.c:7344 socket_process: Registration of 'host2' rejected: 'Registration Refused' from: '10.10.10.82' I was getting a 'Cause Code: 29' INV,POKE,...,REJ but I can't duplicate that level of debugging again in the CLI> on host15 10.10.10.15

Greetings all. I have seen a couple of diagrams on the net that look like a computer set up as an nginx server and more than one mongrel server running at once. What is the advantage of having more than one mongrel server running? Is there a good how-to that explains it very well. thanks shawn -- You received this message because you are subscribed to the Google Groups "Ruby on Rails:

Hey there all if i have this in my gmap_api_key.yml file, production: ABQIAAAASPzYuwhUCPD9Vz98LPOPDhQXfahRKLczKzlJSy9lXc0oaHHYcxTFTjWleXY7LH-N3IAp9ruoqiU43w everything works fine but if i have this ..... production: texaspivot.com: ABQIAAAASPzYuwhUCPD9Vz98LPOPDhQXfahRKLczKzlJSy9lXc0oaHHYcxTFTjWleXY7LH-N3IAp9ruoqiU43w i get this ActionView::TemplateError (texaspivot.com) on line #1 of

I have two seperate systems at two different locations. Each hosts there own voicemail for their phones. I have thought about just having all voicemail on one server. Is the best way to do this just through a dial app? For example, if someone dials 1000 to check voicemail at site A. The dialplan will be something like this on Site A: [context-for-phones-at-one-location] exten =>

I use qualify in sip.conf and need to setup a trigger when asterisk sees it as unreachable, so that I can either drop a call file, or send an email, or both. How can I do that? Thank you

hi, I have 2 asterisks with the following configuration. asterisk server 1 (S1) has an user 9002 asterisk server 2 (S2) has an user 9003 Both users can make call to each other without problem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/9003@S2) or visa versa. Both processes

Hi i'm student and my final project is related to Voip. I have Asterisk almost fully configured. The next step is to accept login of users, that data is in Universitys database which uses ActiveDirectory and also Ldap and Kerberos. It's possible? I don't want authentications in sip.conf, but in other remote database. The problem is i don't have ideas how to start with. I would

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo

Hi all, I upgrade my asterisk from 1.2.11 to 1.4.2 changing my entire dial plan but I have the following errors and I'm not able to call anymore. Do you know what can I have to do? My Asterisk is connected to a patton with a SIP trunk. [Mar 22 10:19:03] WARNING[16462]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call

Hey there all,. i am developing three sites that run on the same code, just different logos, different companies, etc.. i have a config file that i believe is how it should be and it works on webrick, but i get application errors when i try to move it to the apache FCGI . looks like this development: ABQIAAAA8GzGL0c5HitzfNKYM4nk_BQODDn4SSK1uiT7GyxROHR1XtTV_RTBQrvsTtfZDFeAH2vzkOtXunHV9g

I recently upgraded from asterisk 1.2.13 to 1.4.2 and am looking at using the users.conf file to setup my users, before i was using real time SIP which worked fine. However when i create a user in users.conf i am unable to register the user form a softphone, however that same softphone can still register a different the users i currently have setup form the sip.conf from real time. i've

I have set up a script that ensures certain services are up on my Asterisk box (Trixbox 2.0). I would like it to trigger a wake-up call if certain conditions aren't meant. How might I accomplish this from the shell? -- Donovan Niesen Customer Contact Services www.yourccsteam.com

hey there, i have a simple model called key_words.rb the purpose of this is to hold system config data. it also holds the parameters that our system uses to judge the condition of the stuff we monitor. anyway, one of the methods is this def self.get_in_wet_list(status) wet_list = find(:first, :conditions => "name = ''wet_list'' ")

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Hello, Can anybody point me to a sample of popup/tooltip using Prototype? I saw examples for windows and also for tool tip but what I want is a little bit different. I want the same fuctionality which exists at yahoo mail login. https://login.yahoo.com/ -> on the right corner you have a Prevent Password Theft image. On mouse over you see a tooltip. Now the good think in this solution is that

Hi! As the upstream of my DSL-connection is very slow, I'd like my sip-phones to use iLBC to connect to my *. My gateway provider only allows ulaw. Hence, I'd like to use the follwing setup: SIP-phone <--iLBC--> Asterisk <---ulaw----> PSTN-Gateway I get the following error: "Unable to find a codec translation path from ilbc to ulaw" Setup SIP-phone: disallow=all

Hi, first of all i'm sorry for my english. I'm triyng to migrate a Samba 3.6 domain to Samba 4 and I've a question about OU and Winbind: OU affects to something more besides GPO in AD and Winbind?. Because I've moved all users to an OU and all less one (strangely) have dissapear from "getent passwd" and the other SO tools. If i run "wbinfo -u" all users are