search for: bkruse

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2007 Apr 11
10
Nagios asterisk monitoring
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
2007 Apr 24
2
Voicemail on Different Server
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 =>
2007 Apr 12
8
test
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> <HTML><HEAD> <META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"> <TITLE>Message</TITLE> <META content="MSHTML 6.00.2900.3059" name=GENERATOR></HEAD> <BODY> <DIV>&nbsp;</DIV></BODY></HTML>
2007 May 02
2
allowing call to my pabx every 15 minutes
Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this.
2007 May 11
1
'Invalid characters in name' with asterisk-gui
Hi all! Is there a way to asterisk-gui to allow underline (as such cpd_tom) in Names? It allows to [di]enable alphanumeric, but not underline noway. Why such restriction in asterisk-gui if even asterisk users.conf allows (and works fine) it? Thank you, Tom Lobato
2007 Apr 20
2
Queue problems
Hi, I've been configuring AsteriskNOW from the GUI but could it be that the GUI isn't working properly? because when I make a queue and add a few agents, and when I call the queue none of the phones ring. The queue is also configured at "Ringall" I checked the queues.conf file and the settings matched. I also noticed that the agents I made in the GUI, that they were not written
2007 May 01
3
Stanaphone business ok?
I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble.... thanks Todd
2007 May 04
2
question about more than one drop file
hello there all, if i have a script that writes drop files into /var/spool/asterisk/outgoing asterisk picks up the file and initiates the call just fine. however, what is supposed to happen if more than one gets dropped in there within like a second. Will it wait till the first is complete to initiate the second ? Do they dissapear ? thanks shawn -------------- next part -------------- An HTML
2007 Apr 28
7
Two Connected Servers Sound Quailty
Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the