similar to: Empty Queues

Anyone know of an OS/Software inventory tool that supports Linux and Windows that is agentless? Thanks, jlc

I want to include a new context in my exensions.conf I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of follow it?! I have a context [local] that I know zapata.conf points to, I have edited extensions.conf and put in my phone, sip and iax extensions. I want to add an sms context. I understand that all calls go through my [local] context and I have

Hi. I have a problem which I assume would be easy to fix, but I can't find anything about it... I wish to have people dialing my phone, and if it is busy, they are put into a queue. And then I am dialed back when the previous call is finished, and connected to the waiting caller. Easy enough? ----------exten exten => 6,1,Background(salesq-intro); exten => 6,2,Queue(salesq|tT|||300);

Hello everyone, This is is my first email to this group. I'm am writing a small php program to pull some info out of our Asterisk's queue_log. I'm having trouble figuring out what some of the parameters mean. Here's an example: 1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||"Ray Balbin 25" <(716)250-3405> I found a doc that tells me about everything from

Hello everybody, I have started using Asterisk in a call center with ACD. I have noticed something and I wonder if anyone knows whether it is a bug or a feature! I am using Queue application to ring a number of agents that have logged on using AgentCallbackLogin. Now, while an agent receives a call from the Queue they cannot logoff using AgentCallbackLogin. Instead the Agent is asked for

I have this [TRONCAL-SIP] exten=>225/91,1,Answer exten=>225/91,2,Echo exten=>225/91,3,Hangup exten=>225/92,1,Answer exten=>225/92,2,Playback(conf-invalid) exten=>225/92,3,Hangup When I make a call CLI> -- Recv IAM CIC=8 ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=>225,1,Answer Works ok -------------- next part --------------

hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar

Hi all! I'm building a raid box to use for backups, connectivity will be either USB3 or esata. Looking for suggestions on backup software I can use. I know there's rsync, which may be a good solution. I also find backupPC at epel, backintime also at epel, kbackup. DejaDup looks interesting, but none of the repos I'm set up to use shows it being available. some small details: I

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, May 25, 2004 5:30 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3891 - 8 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or

So I'm trying to set up X-Lite for use with Asterisk, and I can't get it to work. I used a PDF I found on Voip-Info to set it up, and it's still not working, I'm using Nufone as the provider... I'll include my sip.cocnf and extentions.conf here. extentions.conf [CODE][outgoing] exten => _1NXXNXXXXXX,1,Dial,IAX2/scheda@NuFone/${EXTEN} [inbound] exten =>

Hello guys, I?m running Asterisk CVS-HEAD-02/01/05-12:22:46 and having a problem with call transfers using the cmds AgentCallBackLogin and AgentLogin First Case (using cmd AgentCallbacklogin): When the incoming call comes and enters the queue, the agent logged in answer the call. But when I try to transfer this call to another agent, the incoming call is dropped. I don?t receive any error

Which kind of transfer do you use? Try using the # transfer. Hope that helps.. Guido Hecken -----Urspr?ngliche Nachricht----- Von: Diego Magalh?es [mailto:diego@redetaho.com.br] Gesendet: Mittwoch, 2. Februar 2005 17:21 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] AgentLogin / AgentCallbackLogin transfer problem Hello guys, I?m running Asterisk CVS-HEAD-02/01/05-12:22:46 and

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter

Hmm i found the problem... I?m using a Grandstream BT100. The transfer just works in a queue if I first acknowledged the call using the # key, and then press the TRANSFER key in the Grandstream. In the asterisk console I receive a: -- SIP/4002-4563 acknowledged Then I can transfer the call... Weird because i?m using ackcall=NO in agents.conf ... Diego Magalh?es diego@redetaho.com.br +55 24

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C

Dear all, I am using the following code to write the plot to an eps format postscript(file="test.eps",horizontal=FALSE)

https://bugzilla.samba.org/show_bug.cgi?id=2455 Summary: rsync --daemon segfaults if "log file = <file>" dir does not exist Product: rsync Version: 2.6.3 Platform: Sparc OS/Version: Solaris Status: NEW Severity: normal Priority: P3 Component: core AssignedTo:

Hi!, I'm trying to set up a Queue (which works fine now :-) Sip clients can login in to the Queue with dialing 91 on there phone. And as soon as there are customers the Queue calls the agents back. I would like that the queue calls the agents also if it's phone is call-forwarded. With agents (sip clients) are added with the following extensions: exten => 91,1,AddQueueMember(myqueue)

Hello all, this may be slightly offtopic :-) I have some Cisco 7940 phones with SIP firmware, connected to an Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g (HorstBox Pro with custom extensions.conf). On some of the phones, two lines are configured, one for business, one for private calls. When forwarding a line to another destination (e.g. to voicemail), we can't use the phone's own

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to