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From: Arnd V. <av...@ne...> - 2006-02-14 13:53:27
|
Hell Jason, thx for the reply! > Perhaps a little explanation will help you here as I think you might have a > slight misunderstanding of what is happening. > > When you 'originate' a call using the asterisk manager, you are sending an > 'originate action' request to the asterisk server, we are using the asterisk java 2.0 API and the "OriginateAction" Class to establish the call. I have to admit i am not the programmer but posting here on his behalf so i dont understand all the jave API centric concepts fully. Simple Question: Is is possible to solve the problem with existing java-classes or do we need to get down to protocol level and issue actions ourselves?! > It sounds like you want to originate call_A using originateAciton in > Manager, and then, if answered and not busy, use dial() in entensions to > connect to the end user, so when they pickup, they're already connected. We want either this or connect to the enduser when the "other side" is "ringing" and not "busy". We only want the user to pickup the fone if the other side is available or picking up. cheers, Arnd |
From: Jason W. <jas...@be...> - 2006-02-13 20:43:24
|
Perhaps a little explanation will help you here as I think you might have a slight misunderstanding of what is happening. When you 'originate' a call using the asterisk manager, you are sending an 'originate action' request to the asterisk server, when answered, control is passed to the server, falling through the extensions that you have in the context that you specified in the originate action... at this point you can parse events that asterisk sends out to your manager instance, or read responses that your manager get's in response to other actions (there are also 'event generating actions'). you have to parse the channel variable in the event to know which channel the event is for in your event handler code. It sounds like you want to originate call_A using originateAciton in Manager, and then, if answered and not busy, use dial() in entensions to connect to the end user, so when they pickup, they're already connected. I have a multi-threaded app which picks up calls from a database of scheduled calls and originates them. up to three numbers are used, so my app remains in contol, and releases it when I know that the call has been answered and accepted. I know it has been accepted because I set a channel variable accepted=yes in the extension and then read it with a manageraction. you have to know the channel to read it's variables... in my case i didn't know from the origination since I'm using IAX and the channel i requested doesn't equal the actual channel used... so in the first priority for the extension, I set a global named by the phone number I sent in the originate action, and the value is the channel... then I can use the channel and go from there. hope this helps. -----Original Message----- From: ast...@li... [mailto:ast...@li...]On Behalf Of Arnd Vehling Sent: Monday, February 13, 2006 3:31 PM To: Ast...@li... Subject: [Asterisk-java-users] originate call without picking up the fone?! Hi There, we are developing a dialer application using the java lib to interface with the asterisk manager protocol. It works fine so far. The only problem we have is that if we use the "originate" command the user is required to pick up the fone _before_ asterisk will originate the call to the desired destination. What we would like to do is to place the call, check if the other end is available (ringing event) and only then let the user pickup the fone. Otherwise we would only display a text message "Destination 'Busy' etc. Does anyone know if/how this is possible? cheers, Arnd ------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Do you grep through log files for problems? Stop! Download the new AJAX search engine that makes searching your log files as easy as surfing the web. DOWNLOAD SPLUNK! http://sel.as-us.falkag.net/sel?cmd=lnk&kid=103432&bid=230486&dat=121642 _______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
From: Arnd V. <av...@ne...> - 2006-02-13 20:18:12
|
Hi There, we are developing a dialer application using the java lib to interface with the asterisk manager protocol. It works fine so far. The only problem we have is that if we use the "originate" command the user is required to pick up the fone _before_ asterisk will originate the call to the desired destination. What we would like to do is to place the call, check if the other end is available (ringing event) and only then let the user pickup the fone. Otherwise we would only display a text message "Destination 'Busy' etc. Does anyone know if/how this is possible? cheers, Arnd |
From: Andre L. M. R. <and...@go...> - 2006-02-13 19:26:09
|
I receive from Asterisk server this response:=20 net.sf.asterisk.manager.response.ManagerResponse: actionId=3D'null'; = message=3D'Originate successfully queued'; response=3D'Success'; = uniqueId=3D'null'; systemHashcode=3D15858385 but the connection don't send for conference room. In the extension I = continue with the external connection made previously. =20 ----- Original Message -----=20 From: Forte, Graham (GFORTE)=20 To: ast...@li...=20 Sent: Monday, February 13, 2006 4:11 PM Subject: [Asterisk-java-users] help in Meetmejoin event!=20 Stephan helped me on this one, try: originateAction =3D new OriginateAction(); originateAction.setChannel("Zap/1-1"); originateAction.setContext("default"); originateAction.setApplication("MeetMe"); originateAction.setData("8200"); //you had no quotes originateAction.setCallerId("Painel <" + 8200 + ">"); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(new Long(30000)); originateAction.setAsync(Boolean.TRUE); //add this -------------------------------------------------------------------------= ----- From: ast...@li... on behalf of = Andre Luiz Martins Rodrigues Sent: Mon 2/13/2006 1:45 PM To: ast...@li... Subject: [Asterisk-java-users] help in Meetmejoin event! Hello everbody. I need of help. In the application that am developing I am finding = the following problem. When I receive a connection, or when I do a = connection in some moments I need to transfer for conference (Event: = MeetmeJoin). However when I perform: =20 originateAction =3D new OriginateAction(); originateAction.setChannel("Zap/1-1"); originateAction.setContext("default"); originateAction.setApplication("MeetMe"); originateAction.setData(8200); originateAction.setCallerId("Painel <" + 8200 + ">"); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(new Long(30000)); Nothing happens. Someone knows to say as I can transfer a connection = for conference? =20 Andr=E9 Luiz Martins Rodrigues Analista de Sistemas - Govoip #VoIP: 3400 0500 Fone: 55 (34) 3214-5500=20 and...@go... |
From: Forte, G. \(GFORTE\) <gf...@ar...> - 2006-02-13 19:12:26
|
Stephan helped me on this one, try: =20 originateAction =3D new OriginateAction(); originateAction.setChannel("Zap/1-1"); originateAction.setContext("default"); originateAction.setApplication("MeetMe"); originateAction.setData("8200"); //you had no quotes originateAction.setCallerId("Painel <" + 8200 + ">"); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(new Long(30000)); originateAction.setAsync(Boolean.TRUE); //add this ________________________________ From: ast...@li... on behalf of Andre = Luiz Martins Rodrigues Sent: Mon 2/13/2006 1:45 PM To: ast...@li... Subject: [Asterisk-java-users] help in Meetmejoin event! Hello everbody. =20 I need of help. In the application that am developing I am finding the = following problem. When I receive a connection, or when I do a = connection in some moments I need to transfer for conference (Event: = MeetmeJoin). However when I perform: =20 =20 originateAction =3D new OriginateAction(); originateAction.setChannel("Zap/1-1"); originateAction.setContext("default"); originateAction.setApplication("MeetMe"); originateAction.setData(8200); originateAction.setCallerId("Painel <" + 8200 + ">"); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(new Long(30000)); =20 Nothing happens. Someone knows to say as I can transfer a connection = for conference? =20 Andr=E9 Luiz Martins Rodrigues Analista de Sistemas - Govoip #VoIP: 3400 0500 Fone: 55 (34) 3214-5500=20 and...@go... |
From: Andre L. M. R. <and...@go...> - 2006-02-13 18:43:42
|
The server response this: AsteriskResponse: net.sf.asterisk.manager.response.ManagerError: = actionId=3D'null'; message=3D'Originate failed'; response=3D'Error'; = uniqueId=3D'null'; systemHashcode=3D29408998 ----- Original Message -----=20 From: Jason Wolfe=20 To: ast...@li...=20 Sent: Monday, February 13, 2006 3:34 PM Subject: RE: [Asterisk-java-users] help in Meetmejoin event! are you actually sending the response? do you also have the = following??? ManagerResponse originateResponse; originateResponse =3D managerConnection.sendAction(originateAction, = 30000); System.out.println("AsteriskResponse: " + = originateResponse.toString()); jason -----Original Message----- From: ast...@li... = [mailto:ast...@li...]On Behalf Of = Andre Luiz Martins Rodrigues Sent: Monday, February 13, 2006 1:45 PM To: ast...@li... Subject: [Asterisk-java-users] help in Meetmejoin event! Hello everbody. I need of help. In the application that am developing I am finding = the following problem. When I receive a connection, or when I do a = connection in some moments I need to transfer for conference (Event: = MeetmeJoin). However when I perform: =20 originateAction =3D new OriginateAction(); originateAction.setChannel("Zap/1-1"); originateAction.setContext("default"); originateAction.setApplication("MeetMe"); originateAction.setData(8200); originateAction.setCallerId("Painel <" + 8200 + ">"); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(new Long(30000)); Nothing happens. Someone knows to say as I can transfer a = connection for conference? =20 Andr=E9 Luiz Martins Rodrigues Analista de Sistemas - Govoip #VoIP: 3400 0500 Fone: 55 (34) 3214-5500=20 and...@go... |
From: Jason W. <jas...@be...> - 2006-02-13 18:37:25
|
sorry, I ment 'sending the action', not 'sending the response' jason -----Original Message----- From: ast...@li... [mailto:ast...@li...]On Behalf Of Jason Wolfe Sent: Monday, February 13, 2006 1:34 PM To: ast...@li... Subject: RE: [Asterisk-java-users] help in Meetmejoin event! are you actually sending the response? do you also have the following??? ManagerResponse originateResponse; originateResponse = managerConnection.sendAction(originateAction, 30000); System.out.println("AsteriskResponse: " + originateResponse.toString()); jason -----Original Message----- From: ast...@li... [mailto:ast...@li...]On Behalf Of Andre Luiz Martins Rodrigues Sent: Monday, February 13, 2006 1:45 PM To: ast...@li... Subject: [Asterisk-java-users] help in Meetmejoin event! Hello everbody. I need of help. In the application that am developing I am finding the following problem. When I receive a connection, or when I do a connection in some moments I need to transfer for conference (Event: MeetmeJoin). However when I perform: originateAction = new OriginateAction(); originateAction.setChannel("Zap/1-1"); originateAction.setContext("default"); originateAction.setApplication("MeetMe"); originateAction.setData(8200); originateAction.setCallerId("Painel <" + 8200 + ">"); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(new Long(30000)); Nothing happens. Someone knows to say as I can transfer a connection for conference? André Luiz Martins Rodrigues Analista de Sistemas - Govoip #VoIP: 3400 0500 Fone: 55 (34) 3214-5500 and...@go... |
From: Jason W. <jas...@be...> - 2006-02-13 18:34:11
|
are you actually sending the response? do you also have the following??? ManagerResponse originateResponse; originateResponse = managerConnection.sendAction(originateAction, 30000); System.out.println("AsteriskResponse: " + originateResponse.toString()); jason -----Original Message----- From: ast...@li... [mailto:ast...@li...]On Behalf Of Andre Luiz Martins Rodrigues Sent: Monday, February 13, 2006 1:45 PM To: ast...@li... Subject: [Asterisk-java-users] help in Meetmejoin event! Hello everbody. I need of help. In the application that am developing I am finding the following problem. When I receive a connection, or when I do a connection in some moments I need to transfer for conference (Event: MeetmeJoin). However when I perform: originateAction = new OriginateAction(); originateAction.setChannel("Zap/1-1"); originateAction.setContext("default"); originateAction.setApplication("MeetMe"); originateAction.setData(8200); originateAction.setCallerId("Painel <" + 8200 + ">"); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(new Long(30000)); Nothing happens. Someone knows to say as I can transfer a connection for conference? André Luiz Martins Rodrigues Analista de Sistemas - Govoip #VoIP: 3400 0500 Fone: 55 (34) 3214-5500 and...@go... |
From: Andre L. M. R. <and...@go...> - 2006-02-13 17:45:43
|
Hello everbody. I need of help. In the application that am developing I am finding the = following problem. When I receive a connection, or when I do a = connection in some moments I need to transfer for conference (Event: = MeetmeJoin). However when I perform: =20 originateAction =3D new OriginateAction(); originateAction.setChannel("Zap/1-1"); originateAction.setContext("default"); originateAction.setApplication("MeetMe"); originateAction.setData(8200); originateAction.setCallerId("Painel <" + 8200 + ">"); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(new Long(30000)); Nothing happens. Someone knows to say as I can transfer a connection = for conference? =20 Andr=E9 Luiz Martins Rodrigues Analista de Sistemas - Govoip #VoIP: 3400 0500 Fone: 55 (34) 3214-5500=20 and...@go... |
From: Gerwin B. <ger...@gm...> - 2006-02-13 17:43:29
|
Hi, has anybody some hints on the following issue: Thx, Gerwin Begin forwarded message: > Hi, > > I've installed Knoppix 4.0.2 on a Dell GX620 P4. > I'm using the JIAXClient on top of IAXClient in a java application > to handle calls. > > On these boxes (3 of them) the sound quality is extreamly crappy, > but the same app running on a win PC is sounding perfect. > > Has anybody got some hints/thoughts where the problem could be? > > Thanks, > Gerwin Begin forwarded message: > From: Steve Kann <st...@st...> > Date: February 13, 2006 6:40:23 PM GMT+01:00 > To: Gerwin Bruner <ger...@gm...> > Subject: Re: [Iaxclient-devel] Knoppix Linux / crappy sound > > Gerwin Bruner wrote: > >> Hi, >> >> the sound is very robotic and breaks up every once in a while. > > > In this case, It could either be a networking issue, or a local > process/thread scheduling issue. On Linux, we can't ask for any > kind of better scheduling priority unless we are running as root. > I'm not that familiar with how Mikael has done all his JNI work, > and if there's a possiblity that that adds some overhead that > affects audio or not. > >> I also just get the iaxclient via: http://www.hem.za.org/ >> jiaxclient/ jiaxclient-current/jiaxtest.html >> and the included autodownload. >> >> for compiling the test app, I just need to get the cvs head >> version, right? > > Yes, I'd get it from sourceforge CVS. See if "testcall" sounds the > same, better, or worse, and also try running testcall as root, and > see if that changes anything. It should give some ideas. > > -SteveK > |
From: Jason W. <jas...@be...> - 2006-02-13 15:25:29
|
John, I did a couple of things to make sure I was making the most of events... You might find it useful to try the following... 1. Use a program like Ethereal to sniff packets being recieved by your server running Manager. This way you see exactly what you're getting. You can filter to only show packets from the asterisk machine, and then you can do a test of the situation that you want to capture and see if asterisk is telling you what you need. 2. Look into 'user defined events' http://www.voip-info.org/wiki/view/Asterisk+cmd+UserEvent 3. grep on the entire asterisk directory for the currently generated events, I'm sure what your looking for would be in in channel.c if it exists, or trust the list at http://www.voip-info.org/wiki/view/asterisk+manager+events 4. if you look at the source for asterisk, you'll see that it's not hard to add events right in the source code and recompile if it's mission critical. 5. as well, Stefan did a great job of laying out Asterisk-java, so you could download the source and add a class for your handler if you create one. Hope this helps. Jason -----Original Message----- From: ast...@li... [mailto:ast...@li...]On Behalf Of John Farnell Sent: Monday, February 13, 2006 1:38 AM To: ast...@li... Subject: [Asterisk-java-users] Call waiting/Busy/Dialing Hi, I am calling setAction(action, handler) on an instance of net.sf.asterisk.manager.DefaultManagerConnection. The 'action' variable is an instance of net.sf.asterisk.manager.action.OriginateAction created as follows :- OriginateAction actionReturn = new OriginateAction(); actionReturn.setChannel(Zap/g0/+ strTelephoneNumber); actionReturn.setExten("200"); actionReturn.setContext("from-internal"); actionReturn.setPriority(new Integer(1)); actionReturn.setTimeout(new Long(30000)); actionReturn.setAsync(new Boolean(true)); I have checked the extensions.config and the from-internal context as per the default install. My goal is to trap events (states?) of dialling/busy/callwaiting/phone number disconnected and ringing in the handleEvent method of my handler class. I am easily able to trap dialling and ringing states via 2 NewStateEvent that are supplied to the handler. However I am not sure which events to interrogate to determine whether a call is any of callwaiting/linebusy/ or line disconnected. Is it possible to use the Manager API to find this information? and if so how can it be done? Will I need to set up a different context on the Asterisk server to achieve this? Any assistance would be greatly appreciated. John Farnell. |
From: John F. <jfa...@no...> - 2006-02-13 06:38:37
|
Hi, I am calling setAction(action, handler) on an instance of = net.sf.asterisk.manager.DefaultManagerConnection. The 'action' variable = is an instance of net.sf.asterisk.manager.action.OriginateAction created = as follows :- OriginateAction actionReturn =3D new OriginateAction(); actionReturn.setChannel(Zap/g0/+ strTelephoneNumber); actionReturn.setExten("200"); actionReturn.setContext("from-internal"); actionReturn.setPriority(new Integer(1)); actionReturn.setTimeout(new Long(30000)); actionReturn.setAsync(new Boolean(true)); I have checked the extensions.config and the from-internal context as = per the default install. My goal is to trap events (states?) of = dialling/busy/callwaiting/phone number disconnected and ringing in the = handleEvent method of my handler class. I am easily able to trap = dialling and ringing states via 2 NewStateEvent that are supplied to the = handler. However I am not sure which events to interrogate to determine = whether a call is any of callwaiting/linebusy/ or line disconnected. Is = it possible to use the Manager API to find this information? and if so = how can it be done? Will I need to set up a different context on the = Asterisk server to achieve this? Any assistance would be greatly appreciated. John Farnell. |
From: Andre L. M. R. <and...@go...> - 2006-02-10 12:34:50
|
Hello everybody, in my application i have one problem. When I try to do = the event meetme with channel zap swims happens.=20 originateAction =3D new OriginateAction();=20 originateAction.setChannel(UniCall/5-1);=20 originateAction.setContext(default);=20 originateAction.setApplication("MeetMe");=20 originateAction.setData(8200);=20 originateAction.setCallerId("Painel <" + extension + ">");=20 originateAction.setPriority(new Integer(1));=20 originateAction.setTimeout(new Long(30000));=20 try{=20 = this.managerConnection.sendAction(originateAction,new = ManagerResponseHandler()=20 Somebody can help me?=20 Andr=E9 Luiz Martins Rodrigues Analista de Sistemas - Govoip #VoIP: 3400 0500 Fone: 55 (34) 3214-5500=20 and...@go... |
From: Andre L. M. R. <and...@go...> - 2006-02-10 12:34:36
|
Hello everybody, in my application i have one problem. When I try to do the event meetme with channel zap swims happens. originateAction = new OriginateAction(); originateAction.setChannel(UniCall/5-1); originateAction.setContext(default); originateAction.setApplication("MeetMe"); originateAction.setData(8200); originateAction.setCallerId("Painel <" + extension + ">"); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(new Long(30000)); try{ this.managerConnection.sendAction(originateAction,new ManagerResponseHandler() Somebody can help me? André Luiz Martins Rodrigues Analista de Sistemas - Govoip #VoIP: 3400 0500 Fone: 55 (34) 3214-5500 and...@go... |
From: Brett S. <bs...@no...> - 2006-02-09 02:25:23
|
I've come across a problem. I have a thread which is using a blocking queue. The queue contains the next action to pass to the manager api. I call sendAction which fails with a timeout (this is as expected). I then call logout which also fails with a timeout (not excepected) finally I loop back to get the next action from the blocking queue only to find that the queue throws an interrupted exception (not expected). Problem 1. The first problem appears to be that you can't log off if an action is in progress! Problem 2. The Default ManagerConnection class seems to be the source of the second problem. The method DefaultResponseHandler.handleResponse(1172) calls thread.interrupt. This appears to happen after the sendAction and logout have timedout. The trigger appears to be that asterisk decides that it can process the original logout request (based on watching the asterisk logs) So I'm assuming the problem is that the handleResponse method assumes that the thread will be in one of the manager api's such as sendAction or logoff in which case presumably the interrupted exception is caught there. The problem in my case is that my thread is asleep waiting for the next action to be queued and the manager api rudely wakes it up to inform it that it has logged off. Regards, Brett |
From: Artem V. L. <te...@of...> - 2006-02-08 06:15:44
|
Hi ! I would like to put on a question about user authentication on Asterisk PBX. I want to store UIDs and passwords for users in MySQL DB instead of keeping this information in sip.conf file and I want to take control over the users login operation. Catching ConnectEvent does not help, because "A ConnectEvent is triggered after successful login to the asterisk server." Any help will be appreciated. // Artem |
From: Olivier B. <Oli...@vo...> - 2006-02-07 15:15:19
|
Stefan, =20 Could you try to install Asteriskwin32 and try this out?=20 =20 Thxs =20 Olivier ________________________________ From: Olivier Brand Sent: Sun 2/5/2006 11:30 PM To: ast...@li...; = ast...@li... Subject: RE: [Asterisk-java-users] AGI Chanell Issue Stefan, =20 Running the script through Eclipse debugger, I can see that the problem = resides in the execution of the answer() operation. =20 The line included in AGIReaderImpl.java: "line =3D socket.readLine()" doe not return anything, so the threads just hangs. =20 On Asterisk console, I can only see the following when "agi debug" is = on: =20 AGI Tx >> agi_network_script: demo.agi AGI Tx >> agi_request: agi://localhost/demo.agi AGI Tx >> agi_channel: SIP/angela-1c3a AGI Tx >> agi_language: en AGI Tx >> agi_type: SIP AGI Tx >> agi_uniqueid: 1139209883.7 AGI Tx >> agi_callerid: "angela" <angela> AGI Tx >> agi_dnid: 1300 AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: tutorial AGI Tx >> agi_extension: 1300 AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >>=20 =20 This is it!! =20 Help please. =20 Olivier =20 =20 =20 =20 =20 ________________________________ From: ast...@li... on behalf of = Stefan Reuter Sent: Sun 1/29/2006 10:08 AM To: ast...@li... Subject: Re: [Asterisk-java-users] AGI Chanell Issue Hi, what version of Asterisk do you use? Please try to enable agi debugging on the CLI and post the output. Maybe that helps trace the problem: srv1*CLI> agi debug AGI Debugging Enabled =3DStefan On Thu, 2006-01-26 at 16:17 -0800, Olivier Brand wrote: > I am trying to implement the example from the tutorial but the example > hangs when a channel method is called (including of course the answer > method). >=20 > 1) I am using the Asterisk version on Windows. > 2) The extension.conf contains the following line: > exten =3D> 1300,1,Agi(agi://localhost/demo.agi) > 3) I am starting the DefaultAGIServer from Eclipse and see the > following logging info at startup: > Jan 26, 2006 4:07:44 PM net.sf.asterisk.util.impl.JavaLoggingLog > info > > INFO: Thread pool started. > > Jan 26, 2006 4:07:44 PM net.sf.asterisk.util.impl.JavaLoggingLog > info > > INFO: Listening on *:4573. > > > 4) I am calling extension 1300 from an Xlite client, the client tries > to connect >=20 > 5) The following logging info shows up in Asterisk PBX console: >=20 > -- Executing AGI("SIP/olivier-6fd9","agi://localhost/demo.agi") in > new stack >=20 > 6) On Eclipse I can see the following logging info: > Jan 26, 2006 4:11:45 PM net.sf.asterisk.util.impl.JavaLoggingLog > info > > INFO: Received connection. > > Jan 26, 2006 4:11:45 PM net.sf.asterisk.util.impl.JavaLoggingLog > info > > INFO: Added mapping for 'demo.agi' to class class > com.vodafone.asterisk.demo.WeatherScript > > Jan 26, 2006 4:11:45 PM net.sf.asterisk.util.impl.JavaLoggingLog > info > > INFO: Begin AGIScript com.vodafone.asterisk.demo.WeatherScript on > AGIServer-TaskThread-0 > > Jan 26, 2006 4:11:45 PM com.vodafone.asterisk.demo.WeatherScript > service > > INFO: HELLO WORLD > > Jan 26, 2006 4:11:45 PM com.vodafone.asterisk.demo.WeatherScript > service > > INFO: net.sf.asterisk.fastagi.impl.AGIChannelImpl@dbe178 > > > 7) The code of the demo is as follow: >=20 > package com.vodafone.asterisk.demo; >=20 > import java.util.logging.Logger; >=20 > import net.sf.asterisk.fastagi.AGIChannel; > import net.sf.asterisk.fastagi.AGIException; > import net.sf.asterisk.fastagi.AGIRequest; > import net.sf.asterisk.fastagi.BaseAGIScript; >=20 > public class WeatherScript extends BaseAGIScript > { > private static Logger logger =3D Logger.getLogger("WeatherScript"); > public void service(AGIRequest arg0, AGIChannel channel) throws > AGIException > { > =20 > logger.info("HELLO WORLD"); > logger.info(channel.toString()); > logger.info(String.valueOf(channel.getChannelStatus())); // > <=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D NEVER GET LOGGED!!! > > channel.answer(); // <=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D DOES = NOT EXECUTE > logger.info("Channel answered"); > //sayDigits("1234"); > //streamFile("demo-congrats"); > channel.hangup(); > } >=20 > } > >=20 |
From:
<rob...@mo...> - 2006-02-07 11:33:48
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Hi! I'm having difficulties in understanding why do my calls f... Currently I'm trying to print the reason by: if (event instanceof OriginateFailureEvent) System.out.println(((OriginateFailureEvent)event).getReason()); But it just prints the value "1" and I can't find a mapping that tells what does it mean? is there a mapping that shows the connection between the numbers and the reason available? like busy, no answer etc. thanks, Robert |
From: <mar...@gm...> - 2006-02-07 08:07:39
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Upps, sorry :( Mar=EDa |
From: Brett S. <bs...@no...> - 2006-02-07 05:00:19
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I'm calling an application (txfax) using an originate. I've managed to monitor events and wait for a hangup event so that I know that the application has completed executing. The question is how do I get the return value from the application. Reading the source of txfax (app_txfax.c) it appears that txfax returns a value of -1 if the call fails. In further reading it appears that many applications increment the priority in some manner. If so can I get the return value be getting the priority and subtracting some base value? If that is the technique then how do I get the priority value? And the code originateAction = new OriginateAction(); originateAction.setChannel(channel.getChannelName() + "/" + Fax.getTelephoneNo(recipient)); originateAction.setContext(channel.getContext()); HashMap<String, String> variables = new HashMap<String, String>(); Configuration config = Configuration.getInstance(); variables.put("MaxRetries", "1"); variables.put("RetryTime", new Integer(config.getRetryTime()) .toString()); variables .put("WaitTime", new Long(config.getWaitTime()).toString()); originateAction.setVariables(variables); originateAction.setApplication("txfax"); String data = fax.getTiffFile().getFile().getAbsolutePath() + "|caller"; if (config.isDebugTxFax()) data += "|debug"; originateAction.setData(data); originateAction.setPriority(new Integer(config.getPriority())); originateAction.setTimeout(new Long(config.getTimeOut())); Any help would be appreciated. Regards, Brett |
From: Richard H. <ri...@ha...> - 2006-02-06 23:15:33
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Mar=EDa Ch=F3liz wrote: > Yes, it was that !!!!!! Thanks a lot for your help, you have been so=20 > kind, > > = =20 > Mar=EDa=20 Can I just say that these types of questions are better suited to the=20 asterisk-users mailing list not asterisk-java-users. This list is for questions relating to asterisk-java only please. Regards Richard |
From: <mar...@gm...> - 2006-02-06 15:50:07
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Yes, it was that !!!!!! Thanks a lot for your help, you have been so kind, Mar=EDa |
From: Mattias M. <ma...@wx...> - 2006-02-06 15:14:41
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Maybe I should show you what I found on the matter. =3D) =20 http://www.vmware.com/support/kb/enduser/std_adp.php?p_faqid=3D1420 =20 Good Luck! ________________________________ Fr=E5n: ast...@li... = [mailto:ast...@li...] F=F6r Mar=EDa = Ch=F3liz Skickat: den 6 februari 2006 15:11 Till: ast...@li... =C4mne: Re: [Asterisk-java-users] Reproducing sounds in Asterisk with a = bad quality =20 =20 Thank you so much; i'll check it now. The problem is that the sound = plays slowly and with a some noise in the background. I hope it's the = same case as yours,...so my problem will be worked out! Thanks!, |
From: <mar...@gm...> - 2006-02-06 14:11:01
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Thank you so much; i'll check it now. The problem is that the sound plays slowly and with a some noise in the background. I hope it's the same case a= s yours,...so my problem will be worked out! Thanks!, |
From: Mattias M. <ma...@wx...> - 2006-02-06 12:41:13
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Hi, =20 I'm not sure that your problem is the same as mine, but does the sound = also play really slow? I had this problem when configuring an Asterisk on a virtual server and = found out that the virtual machine had problem with its internal clock = and therefore didn't play the sounds correctly. Check if the clock on = the server runs really slow. That is probably the problem. =20 // Mattias Malmquist =20 ________________________________ Fr=E5n: ast...@li... = [mailto:ast...@li...] F=F6r Mar=EDa = Ch=F3liz Skickat: den 6 februari 2006 13:13 Till: ast...@li... =C4mne: [Asterisk-java-users] Reproducing sounds in Asterisk with a bad = quality =20 Hi, I am running asterisk in a Virtual Machine, and I don't why the .wav = files it plays sound really bad,...with some noise in the background. I = generate them with : =20 sox SourcepathFile.* -r 8000 -c 1 DestinypathFile.wav or echo $cadena | text2wave -o $1 -otype riff -f 8000 -scale 6 - In another machine (hardware not virtual ) with Asterisk 1.07 it = reproduces sounds perfectly . The virtual machine has Asterisk = 1.2.1....can this be happening because of the fact that the machine is = virtual??? Thanks in advance, --=20 Mar=EDa=20 |