similar to: ANN: Asterisk-Java 1.0.0.M3 Released

Hi, We've just finished adding support for writing AGI scripts in a variety of popular scripting languages to Asterisk-Java. The FastAGI server in Asterisk-Java allows you to move your AGI scripts to a dedicated server and increases performance by eleminating the need to start the language interpreter for each request. Our current snapshot release includes an AGI demo in Groovy, JavaScript

With Asterisk-Java the proposed solution to connect to multiple Asterisk servers is to create multiple AsteriskManagerConnection obeject. Each ManagerConnection handles its own thread so there is no need for custom thread handing code. All you have to do is to make sure is the EventListener objects you pass to these connections synchronize access to shared data (if there are such accesses). I

robert home wrote: > does any one know what happened to www.asterisk-java.org > or when it'll be back We had problems with the IN NS records at PSI. The problem is fixed now though it might still take a few hours for the changes to propagate. I am sorry for any inconvinience this outage may have caused and have provided a bonus article on Local/ channels to say sorry. The article

Hi, BRIstuff contains two bugs in its implementation of the Redirect manager action: 1. If the property ExtraUnqiueId is used, the Priority property is used to redirect the extra channel (instead of ExtraPriority) 2. If the property ExtraChannel is used, 0 is used to redirect the extra channel regardless of the Priority and ExtraPriority properties. A patch for manager.c is available at

Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel

Hello, Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager and it makes the CLI output very hard to follow because of the " == Parsing '/etc/asterisk/manager.conf': Found". (Yes, Found! manager.conf was there 3 seconds ago, guess what it's still there.) There is a very old feature request about this

Hello all, A client of us, needs a queue monitoring system. In realtime he needs to now the PRI status, the agents logged in and logged out, the number of received calls by agent, ....,etc. I am not a call center specialist and i want to find a call center software to offer to my client that fits his needs. I need a monitoring solution for incomming and outgoing calls and a queue management

Hi - Am I allowed to have multiple managers logged in with the same manager username at the same time? I'm referring to the id names in manager.conf. I expect so, but just want to check to help in troubleshooting a problem. thanks -todd-

Hi, I'm wondering whether or not I should go for ODBC or IMAP voicemail storage. Before diving into details, I would be very pleased to get input form others. 1. With IMAP, is it necessary to save a copy of voicemails in /var/log files so that a user can still listen to his (or her) own voicemails with his own hardphone ? 2. How then, can you make sure to skip non-voice mails stored in the

Dear all, I've noticed that when I have a phone registered in Asterisk, and then I register another phone with the same user, the "sip show peers" in the CLI shows that Asterisk replaced the IP of the first phone by the IP of the last one registered for that user. Consequently, if someone calls that user, only the last phone rings!! How may I configure Asterisk to be able to

Thanks Stefan for input. I know that there is a "hangup" action in Asterisk Manager API. I am looking for "hold and retrive" commend. I search google and find that redirecting to parkslot can work. If I have a PSTN call connecting to Asterisk and then to a SIP extension, there are two connections here. If I redirect one channel to parkslot, another channel will automatically

CentOS Errata and Bugfix Advisory 2012:1239 Upstream details at : https://rhn.redhat.com/errata/RHBA-2012-1239.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) i386: 94ffbefb4195c8e3b668266bdb66a16daab250d4a3041c122d16dbfe693b12cd slf4j-1.5.8-8.el6.noarch.rpm ce90eb72d617088fff67f912f310a6bed41a2f063f17cedaa818b72f7cb4ff3a

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CentOS Errata and Security Advisory 2018:0592 Important Upstream details at : https://access.redhat.com/errata/RHSA-2018:0592 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 733630907981b82d45bd40cf4d3f113ff2193a4fdf1e293818669a707b739189 slf4j-1.7.4-4.el7_4.noarch.rpm

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Hi, I'd like to use the convenience of apt packaging, but debian sarge's default asterisk is something like 1.0.7. Are there any apt repositories which provide newer versions of the software? Thanks! -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hi all, I want to be able to see the status of my Agents on a web interface. I have no idea how to do so. I have found a few sample script to communicate with queues manager to view queues.But I couldn't find any on viewing the agent status. Could anybody give me a clue? Regards, Pim

Our windows users are looking for a simple application to permit dialling and transfer from Windows desktop (or web page). I've looked at everything mentioned in the WIKI, and most are either not appropriate, or are not maintained any longer. I've used Flash Operator Panel, and quite like it, but I don't believe there is a way to have a per-user view (so people can only manage

These are common functions. Why "Asterisk Manager" doesn't provide commands to hold and retrive an active channel? If it must be implemented by AGI, could anyone give a direction or steps? Thanks in advance, James -------------- next part -------------- An HTML attachment was scrubbed... URL:

i've seen that opening a socket on the asterisk server i can originate a call from one extension to another in a specific context. Is it possible to transfer an existing call from the extension ... SIP/xxx to another extension in a specific context? thanks