similar to: [patch] fix for redirect manager action with BRIstuffed Asterisk

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, 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

Hi, We've just released milestone 3 of Asterisk-Java 1.0.0. Next to a few bug fixes this new milestone makes Asterisk-Java OSGi compliant and adds support for the modern SLF4J logging framework. Have a look at http://blogs.reucon.com/asterisk-java/2009/10/19/asterisk_java_1_0_0_m3_released.html =Stefan -- reuter network consulting Neusser Str. 110 50670 Koeln Germany Telefon: +49 221

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

B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to determine the real cause of failure. Also, there's no way to link between the channel that was

Hi! I'm building an application which allows to dial via the Asterisk Manager Interface using the originate command. There should be an optional conferencing feature. The manager commands are basically: --------------------------------- action: login username: sdjklgdsjg secret: xxx events: on action: originate callerid: 3847438609 priority: 1 exten: 4068439865 async: 1 context: out

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

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 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, 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

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

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:

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

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

i am developing a web application to manage callcenter, i will shortly release it on sf.net this is my problem: i will grant to users the possibility to transfer calls to other users using a web interface, for example if SIP/200 is talking with SIP/400 who wants to transfer the call to SIP/500 i use this commands with manager API: Action: Redirect\r\n Channel: SIP/200-sads\r\n ExtraChannel:

Hi All, I use the Asterisk Manager Interface to redirect the channels. There have two channels : SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456) SIP/612-5456 s@macro-monitor:10 Up Dial(SIP/0882@voip_out Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel: