similar to: queues - limiting ringing calls to queue members

Hello fellow asterisk users, I'm trying to use feature application defined in application map. it's defined as follows: lbxvml => 1,self/caller,Macro,Jump2Voicemail It's working properly when called party answers the call, but I'd like to have feature usable while call is still ringing in queue but this just does not work.. Is this a bug or feature? Is there a way to have

Hi, I'm trying to debug crashing server, and I'm starting to suspect tftp server as one of possible culprits. The box is quad core Xeon with 4GB of memory serving as Asterisk PBX + some additional services (reporting, IS integration etc). Among others, it's also serving as provisioning box for over hundred of SPA504 phones. Phones are checking configuration files every 60s meaning

Hi fellow libvirt users, I'd like to ask, whether somebody possibly dealt with similar problem we're hitting.. Some of libvirt VM operations (ie fs freeze) are prone to hang for long time, in case the guest agent is in some bad state.. My question is, if it's possible to set some timeout for such operations, or we have to deal with it ie with separate thread and some timers? we're

Hi, plese have you some idea for this problem? thanks, Lukas -- ---------- Forwarded message ---------- Date: Wed, 30 Sep 2009 23:09:59 +0200 (CEST) From: extmaillist at linuxbox.cz To: pilsl at goldfisch.at, Volker.Lendecke at SerNet.DE Cc: samba at lists.samba.org X-Spam-Score: -1.0 (), 4 required Subject: Re: [Samba] user cannot logon to domain although log says "auth

Hi, I'm trying to convert some quicktime files into Ogg/Theora using the Xiph QuickTime Components version 1.8. The end goal is to stream these files using apache httpd and the cortado applet (and hopefully the embedded players that will be in Firefox soon). Using iMovie DV on OS X 10.4.11 (intel) I'm able to export as an ogg movie, however the resultant file has problems with audio on

Greetings, I have a very basic equal-weight ring-all queue set up in queues.conf: [sales-queue] ;music = default strategy = ringall periodic-announce-frequency = 20 announce-holdtime = no timeout = 15 maxlen = 0 member => SIP/1xxxxxxxxxx at junction_networks,1 member => SIP/1xxxxxxxxxx at junction_networks,1 member => SIP/dude,1 member => SIP/homie,1 member => SIP/fellow,1 But

Dear list, I want to avoid sending calls to a queue when it is full. From the fact that 'maxlen' must be at least 1 (I wish it could be zero but it isn't) I'd like to know if there's a way to do it. Setting the Queue() timeout to a little value is not the most suitable option. I'm using asterisk 1.4.21 but I don't know if there are some options available on release 1.8

I spoke too soon in my last email. My DNS is not working. The host commands from the AD DC how-to all fail with NXDOMAIN. nslookup says server can't find nikola.ozco.home. Basically the same message. Here is a dump of my many tries. ----------------------------------------------------------------------------------------------------------------- nikola ~ # export

Hello, I'm using MySQL realtime Call Queues (table /queues/ and table /queue_members/). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy /lineair /and I put the members into the table /queue_members/ in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout |

Hi All, Couple easy questions: 1. How do I view what environment my app is running under (i.e production / testing / development) - I''m not sure what RAILS_ENV is set at, how to I take a look at it? 2. Somehow (?) I have turned logging "off". I''d like to have it back on :s I have files called production.log, test.log, server.log, development.log in /myapp/log/ but

I''m using Restful Authentication, and the code to create a user is pretty straight forward - there is a before_save action and a before_create action: before_save :encrypt_password before_create :make_activation_code But for some reason when I try to create a user programmatically in the controller like this: User.new(:email =>

In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd like to do something possibly unique with the formatting of extensions in my dial plan, and am having trouble. We use VoicePulse connect, which gives us local DID for inbound and outbound calls (even though DTMF tones are not working in Voice Pulse Connect at the moment). To dial local numbers, you have to

Hi I've got the error "cannot allocate a new connection -- maximum of 16 connections already opened" after I tried to create a new connection to a database. However, the reason ist, that i did not disconnect previous connections.... I don't know the name of this connections. How can I disconnect this "unknown" connections and drivers? if I delete all objects, the

Hi all, i have a problem: i have an adsl modem that is connected to internet. I can''t manage this modem. Between my PC and the modem i have a linux firewall that make the NAT and the traffic shapping. I have create a script that limit the bandwidth of the "external" interface of the firewall so i can manage my bandwidth for my internet application. The problem is that i need to

On Thursday, January 15, 2015 06:55 AM, Nikola Smiljanic wrote: > Was there an rc3 that I missed? > It's not tagged if there was. I've tested and uploaded 3.5.1 final for x86_64-ubuntu-14.04. I think the rc2 build required a newer libstdc++ than exists on Ubuntu 14.04, but the final build should be good. Ben

I upgraded to the latest HEAD version of asterisk, and all IAX calls started sounding choppy. It was suggested on the IRC channel that I go back to asterisk -stable to determine if that fixes it. Is downgrading as simple as upgrading? Because now, -stable builds fine, but I get an error on the asterisk console when starting, something about "ast_get_txt" not found. Recompiling and

Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, Can any one help to install my jar file automatically, When I tried to install /usr/bin/java -Xmx256m -jar /gx/mnt/software/Vidispine/Components/glassfish-installer-v2.1.1-b31g-linux.jar,its is asking for confirmation like Accept or Decline? [A,D,a,d]. So how can I pass answer with command. Please find my puppet code below exec { ''glassfishInstaExe'':

Hi Nik, Flushing MDB can only be done when we are managing it per station not per port. For that we need to have MCAST_TO_UCAST, EHT and FAST_LEAVE. Here one more point is - some vendors may offload MCAST_TO_UCAST conversion in their own FW to reduce CPU. So, the best way is to have MCAST_TO_UCAST enabled and MDB will become per station, so we can delete MDB on disconnect. Shall, I create one

Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. So I want to ask if I'm right or wrong: I set up only a queue , is to say operatorq, where the only member