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From: Jan E. <jan...@pl...> - 2005-08-22 06:45:05
|
On Monday 22 August 2005 09:10, Stefan Reuter wrote: > > Which version of Asterisk does the CVS version target? I'm still stuck > > with Astersik 1.0.7 and we're not planning on upgrading before the next > > version stabilizes. But, I'd love to give it a spin. :) > > Though CVS-HEAD adds support for the features of the upcoming Asterisk > 1.2 release it still supports Asterisk 1.0.x. Then I will happily test it. I don't really expect there to be any problems, as normal with this project... It's kinda boring when all the bugs are in my code. :) -- Jan Ekholm jan...@pl... |
From: Stefan R. <sr...@re...> - 2005-08-22 06:10:41
|
> Which version of Asterisk does the CVS version target? I'm still stuck wi= th=20 > Astersik 1.0.7 and we're not planning on upgrading before the next versio= n=20 > stabilizes. But, I'd love to give it a spin. :) Though CVS-HEAD adds support for the features of the upcoming Asterisk 1.2 release it still supports Asterisk 1.0.x. =3DStefan |
From: Jan E. <jan...@pl...> - 2005-08-22 06:00:14
|
On Monday 22 August 2005 02:09, Stefan Reuter wrote: > On Fri, 2005-08-19 at 13:32 +0300, Jan Ekholm wrote: > > On Friday 19 August 2005 13:27, Stefan Reuter wrote: > > > Yep, DefaultAsteriskManager should take care of this. > > > I'll have a look at it and fix it. Thanks for the hint. > > > > Should just be to have callbacks for Join- and LeaveEvent and add/remove > > the channel. A few lines of code. I do it manually now and that works > > well too. > > Ok, its fixed in latest CVS-HEAD. > Would be cool if you could test it. > For further comments you can access the associated issue at > http://jira.reucon.com/browse/AJ-1 Which version of Asterisk does the CVS version target? I'm still stuck with Astersik 1.0.7 and we're not planning on upgrading before the next version stabilizes. But, I'd love to give it a spin. :) -- Jan Ekholm jan...@pl... |
From: Stefan R. <sr...@re...> - 2005-08-21 23:09:59
|
On Fri, 2005-08-19 at 13:32 +0300, Jan Ekholm wrote: > On Friday 19 August 2005 13:27, Stefan Reuter wrote: > > Yep, DefaultAsteriskManager should take care of this. > > I'll have a look at it and fix it. Thanks for the hint. >=20 > Should just be to have callbacks for Join- and LeaveEvent and add/remove = the=20 > channel. A few lines of code. I do it manually now and that works well to= o.=20 Ok, its fixed in latest CVS-HEAD. Would be cool if you could test it. For further comments you can access the associated issue at http://jira.reucon.com/browse/AJ-1 =3DStefan |
From: Jan E. <jan...@pl...> - 2005-08-19 10:32:05
|
On Friday 19 August 2005 13:27, Stefan Reuter wrote: > Yep, DefaultAsteriskManager should take care of this. > I'll have a look at it and fix it. Thanks for the hint. Should just be to have callbacks for Join- and LeaveEvent and add/remove the channel. A few lines of code. I do it manually now and that works well too. -- Jan Ekholm jan...@pl... |
From: Stefan R. <sr...@re...> - 2005-08-19 10:27:44
|
Yep, DefaultAsteriskManager should take care of this. I'll have a look at it and fix it. Thanks for the hint. =3DStefan |
From: Jan E. <jan...@pl...> - 2005-08-19 10:14:37
|
Hi, When using the queues in Asterisk-Java I noticed that channels aren't populated into queues when they arrive. I mean, if I call a queue I'll get a JoinEvent, and I always see that the queues are empty, they contain no channel entries. Only entries that are in queues when the application starts up are populated into queues (via QueueEntryEvent), as those seem to be handled ok. Is Asterisk-Java meant to internally handle joining and leaving channels for queues, or should I take care of it myself? I can just call Queue.addEntry() or removeEntry() in a suitable callback, but it feels a bit "wrong" to mess with Asterisk-Java internal objects in this way. I think it's a bit misleading to take care of some of the events and not all. :) -- Jan Ekholm jan...@pl... |
From: Stefan R. <sr...@re...> - 2005-08-18 19:55:38
|
On Thu, 2005-08-18 at 16:57 +1000, Samant Nagpaul wrote: > If there a way to switch off Log4JLogger? If you don't include log4j on your classpath it is not used. If you have log4j on your classpath you can exclude asterisk-java's clases from logging by configuring log4j accordingly. =3DStefan |
From: Samant N. <sa...@ai...> - 2005-08-18 18:06:45
|
If there a way to switch off Log4JLogger? Samant |
From: Jan E. <jan...@pl...> - 2005-08-17 12:16:50
|
On Wednesday 17 August 2005 14:27, Stefan Reuter wrote: > > Or, (after thinking a bit more) am I right to assume that when I've > > initialized a ManagerConnection and assigned it to DefaultAsteriskManager > > using setManagerConnection(), that the connection is ok at that point? I > > see > > that DefaultAsteriskManager.initialize() sends off CommandAction:s, so > > probably I can do it as well at this point. > > Ah ok if that is the question the answer is "Yes". Yes, that was the question. :) Thanks for the answer. -- Jan Ekholm jan...@pl... |
From: Stefan R. <sr...@re...> - 2005-08-17 11:27:29
|
> Or, (after thinking a bit more) am I right to assume that when I've > initialized a ManagerConnection and assigned it to DefaultAsteriskManag= er > using setManagerConnection(), that the connection is ok at that point? = I > see > that DefaultAsteriskManager.initialize() sends off CommandAction:s, so > probably I can do it as well at this point. Ah ok if that is the question the answer is "Yes". |
From: Jan E. <jan...@pl...> - 2005-08-17 11:09:59
|
On Wednesday 17 August 2005 13:58, Stefan Reuter wrote: > Hi Jan, > > ConnectEvents are not dispatched because they don't represent "real" > Asterisk events. We could easily change that but I am not sure if that > leads to some confusion. > I recently added getVersion() and getVersion(String) to > DefaultAsteriskManager. These two methods also use a CommandAction to > query Asterisk state. I implemented them using lazy initialization so i > check if the corresponding attribute is null or not and if it is null i > fetch the stuff from Asterisk, otherwise I just return the cached result. > Would that work for you, too? Hi, I don't really see how that example would help me with my problem. I'm interested in known when I can send my own CommandAction:s to get some initial data (users, conferences etc) and populate my UI. The ConnectEvent would be handy as I'd then know that I'm now connected ok. Or, (after thinking a bit more) am I right to assume that when I've initialized a ManagerConnection and assigned it to DefaultAsteriskManager using setManagerConnection(), that the connection is ok at that point? I see that DefaultAsteriskManager.initialize() sends off CommandAction:s, so probably I can do it as well at this point. I'll have to test... -- Jan Ekholm jan...@pl... |
From: Stefan R. <sr...@re...> - 2005-08-17 10:58:40
|
Hi Jan, ConnectEvents are not dispatched because they don't represent "real" Asterisk events. We could easily change that but I am not sure if that leads to some confusion. I recently added getVersion() and getVersion(String) to DefaultAsteriskManager. These two methods also use a CommandAction to query Asterisk state. I implemented them using lazy initialization so i check if the corresponding attribute is null or not and if it is null i fetch the stuff from Asterisk, otherwise I just return the cached result. Would that work for you, too? =3DStefan P.S. the relevant code passage is available at http://asterisk-java.sourceforge.net/xref/net/sf/asterisk/manager/Default= AsteriskManager.html#262 |
From: Jan E. <jan...@pl...> - 2005-08-17 09:27:05
|
Hi, After refactoring my initial test application (a big ugly kludge) I noticed that I don't actually get ConnectEvent:s when Asterisk-Java has connected to the Asterisk server. This event would be really nice, as I need to retrieve some extra info from Asterisk (using CommandAction) as soon as the connection is ok. I see this logged by Asterisk-Java: received: net.sf.asterisk.manager.event.ConnectEvent: dateReceived=Wed Aug 17 12:11:06 EEST 2005; systemHashcode=9564165 received: net.sf.asterisk.manager.event.StatusCompleteEvent: dateReceived=Wed Aug 17 12:11:06 EEST 2005; systemHashcode=28291271 received: net.sf.asterisk.manager.event.QueueParamsEvent: dateReceived=Wed Aug 17 12:11:06 EEST 2005; systemHashcode=25346354 received: net.sf.asterisk.manager.event.QueueMemberEvent: dateReceived=Wed Aug 17 12:11:06 EEST 2005; systemHashcode=11737975 received: net.sf.asterisk.manager.event.QueueParamsEvent: dateReceived=Wed Aug 17 12:11:06 EEST 2005; systemHashcode=13645178 The events are handled internally and not dispatched to event listeners until both channels and queues are internally initialized. An easy fix is to just enqueue ConnectEvents onto the queue of events that get dispatched once all is initialized. Or maybe there is a better way to perform some extra setup as soon as a connection is made? Currently my best bet (without modifying Asterisk-Java) is to just wait for any event to arrive and at that point do the setup once. -- Jan Ekholm jan...@pl... |
From: Peter H. <pe...@li...> - 2005-08-10 19:57:22
|
>> Right, but the escape digits only allow for a single dtmf input, correct? > > yes, but if you detect that the user escaped you can read the missing > digits via waitForDigit > I'll give that a try. If there are still any missed dtmf, it'd probably be an asterisk issue. Thanks again, Peter |
From: Stefan R. <sr...@re...> - 2005-08-10 19:38:50
|
> Right, but the escape digits only allow for a single dtmf input, correc= t? yes, but if you detect that the user escaped you can read the missing digits via waitForDigit |
From: Peter H. <pe...@li...> - 2005-08-10 19:13:49
|
>>> Before switching to AGI script, I simply relied on the extension being >>> inputted. i.e. a user hitting '123' would be routed to exten => >>> 123,1,... >>> Now, I'm trying to use the Get Data to check for 123. >>> >>> Another thing I should mention is that I'm stringing a couple of Get >>> Data >>> commands together to play different wav files. Would this cause any >>> problems? >>> >> >> Hmm.. I'm also setting the timeout to 1 ms for the files that I'm >> streaming >> that are immediately proceeded by another streaming file. Would this >> cause >> any issues? > > I wouldn't use Get Data in that case but rather just Stream File. You can > provide escape digits to stream file so you can interrupt it. I think > thats a cleaner solution. Right, but the escape digits only allow for a single dtmf input, correct? Peter |
From: Stefan R. <sr...@re...> - 2005-08-10 09:01:40
|
>> Before switching to AGI script, I simply relied on the extension being >> inputted. i.e. a user hitting '123' would be routed to exten =3D> >> 123,1,... >> Now, I'm trying to use the Get Data to check for 123. >> >> Another thing I should mention is that I'm stringing a couple of Get >> Data >> commands together to play different wav files. Would this cause any >> problems? >> > > Hmm.. I'm also setting the timeout to 1 ms for the files that I'm > streaming > that are immediately proceeded by another streaming file. Would this > cause > any issues? I wouldn't use Get Data in that case but rather just Stream File. You can provide escape digits to stream file so you can interrupt it. I think thats a cleaner solution. =3DStefan |
From: Peter H. <pe...@li...> - 2005-08-10 08:46:31
|
> >> Hi Peter, >> >>> Occassionally, when executing the script, dtmf input is recognized, and >>> the >>> AGIReply will always list the response as null (timeout). This will >>> then >>> happen for the duration of the call, even if subsequent Get Data >>> commands >>> are executed. >> >> That sounds odd. However if this only happens from time to time I am >> almost sure it is not an Asterisk-Java problem. >> What type of channel do you use in this scenario? (IAX2, SIP, Zap, ..?) >> > > I'm using IAX2 channels, but it seems sporadic... I just wish I could > reproduce the error so I could get some more details. I've run 30 tests > or so in a row w/o hitting the error again, but it has happened to me > about 3 times total on 3 different providers on 3 different days. > >>> This put the provider and asterisk at the top of my list. However, I >>> hadn't >>> had that problem with any of my providers prior to switching over to >>> using >>> AGI script. >> >> Did you process DTMF prior to switching? > > Before switching to AGI script, I simply relied on the extension being > inputted. i.e. a user hitting '123' would be routed to exten => 123,1,... > Now, I'm trying to use the Get Data to check for 123. > > Another thing I should mention is that I'm stringing a couple of Get Data > commands together to play different wav files. Would this cause any > problems? > Hmm.. I'm also setting the timeout to 1 ms for the files that I'm streaming that are immediately proceeded by another streaming file. Would this cause any issues? Peter |
From: Peter H. <pe...@li...> - 2005-08-10 08:32:32
|
----- Original Message ----- From: "Stefan Reuter" <sr...@re...> To: <ast...@li...> Sent: Wednesday, August 10, 2005 1:04 AM Subject: Re: [Asterisk-java-users] Re: Receiving DTMF digits via FastAGI > Hi Peter, > >> Occassionally, when executing the script, dtmf input is recognized, and >> the >> AGIReply will always list the response as null (timeout). This will then >> happen for the duration of the call, even if subsequent Get Data commands >> are executed. > > That sounds odd. However if this only happens from time to time I am > almost sure it is not an Asterisk-Java problem. > What type of channel do you use in this scenario? (IAX2, SIP, Zap, ..?) > I'm using IAX2 channels, but it seems sporadic... I just wish I could reproduce the error so I could get some more details. I've run 30 tests or so in a row w/o hitting the error again, but it has happened to me about 3 times total on 3 different providers on 3 different days. >> This put the provider and asterisk at the top of my list. However, I >> hadn't >> had that problem with any of my providers prior to switching over to >> using >> AGI script. > > Did you process DTMF prior to switching? Before switching to AGI script, I simply relied on the extension being inputted. i.e. a user hitting '123' would be routed to exten => 123,1,... Now, I'm trying to use the Get Data to check for 123. Another thing I should mention is that I'm stringing a couple of Get Data commands together to play different wav files. Would this cause any problems? Peter |
From: Stefan R. <sr...@re...> - 2005-08-10 08:04:54
|
Hi Peter, > Occassionally, when executing the script, dtmf input is recognized, and > the > AGIReply will always list the response as null (timeout). This will th= en > happen for the duration of the call, even if subsequent Get Data comman= ds > are executed. That sounds odd. However if this only happens from time to time I am almost sure it is not an Asterisk-Java problem. What type of channel do you use in this scenario? (IAX2, SIP, Zap, ..?) > This put the provider and asterisk at the top of my list. However, I > hadn't > had that problem with any of my providers prior to switching over to us= ing > AGI script. Did you process DTMF prior to switching? =3DStefan |
From: Peter H. <pe...@li...> - 2005-08-10 07:01:38
|
I'm getting an error with using sendCommand(GetDataCommand). I'm not sur= e=20 if this is an asterisk problem or if it's a asterisk-java problem or if i= t's=20 a provider problem. Occassionally, when executing the script, dtmf input is recognized, and t= he=20 AGIReply will always list the response as null (timeout). This will then= =20 happen for the duration of the call, even if subsequent Get Data commands= =20 are executed. This put the provider and asterisk at the top of my list. However, I had= n't=20 had that problem with any of my providers prior to switching over to usin= g=20 AGI script. Any ideas what might be causing this? Peter ----- Original Message -----=20 From: "Stefan Reuter" <sr...@re...> To: <ast...@li...> Sent: Wednesday, July 27, 2005 3:11 AM Subject: [Asterisk-java-users] Re: Receiving DTMF digits via FastAGI >> A question about the FastAGI stuff in general. Many of the functions >> provided by AbstractAGIScript allow DTMF input to be received from the >> server. >> Is there a way to catch multiple dtmf input instead of a single=20 >> character? >> i.e. I can wait for any digit from 0-9, but I can't wait for '10'.. > > Most of Asterisk's AGI commands can be interrupted by pressing a DTMF=20 > digit. > Which - if any - DTMF digits should cause an interruption can be specif= ied > via the escapeDigits parameter of many of AbstractAGIScript's methods. = The > singe character that those methods return corresponds to the digit that > has been pressed to interrupt the command. > There is no way to say "Abort this command only if user presses DTMF di= git > 2 after he pressed DTMF digit 1". > > If you want to read DTMF digits for further processing, you can use the > GetDataCommand. See > http://asterisk-java.sourceforge.net/apidocs/net/sf/asterisk/fastagi/co= mmand/GetDataCommand.html > The corresponding getData() methods are available in Asterisk-Java > CVS-HEAD, but you can as well use the GetDataCommand with Asterisk-Java > 0.1 vial channel.sendCommand() > > =3DStefan > > > > ------------------------------------------------------- > SF.Net email is sponsored by: Discover Easy Linux Migration Strategies > from IBM. Find simple to follow Roadmaps, straightforward articles, > informative Webcasts and more! Get everything you need to get up to > speed, fast. http://ads.osdn.com/?ad_idt77&alloc_id=16492&op=CCk > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users >=20 |
From: Rick H. <ri...@ha...> - 2005-08-05 18:47:32
|
Hi Andrew, I like the idea of this plugin too, I was going to be implementing something along the same lines (and have done previously, but not in as elegant manner). Could you see if kopete messenger would also be interested in supporting your feature? Nice work, Rick Andrew Wright wrote: > Stefan, > > It is cool that you have had the chance to check this out and it is > mostly working for you :) > > Please see my comments below: > > On Aug 3, 2005, at 5:23 PM, Stefan Reuter wrote: > >>> The 1.0 beta1 of the Asterisk-IM Jive Messenger XMPP (jabber) server >>> plugin has been released today. >>> >> >> hey, that looks great. >> >> now we must get the mainstream jabber clients to support the protocol >> extension... i just tried it with the xml console of psi and made it >> dial. > > > People from Trillian, Gaim, and Pandian have all committed to > supporting the plugin. > Hopefully we will see something from them in the near future. I am > sure others will follow suit afterwards. > > >> >> how do i get the events and presence updates? > > > The events should be sent as long as your asterisk manager user has > the correct perms. Your manager should have the following privleges: > read/write = call,command > > > You won't see the presence change on your client, though other users > can see your status changes. This is because Presence packets aren't > sent to the user who changed their presence. Clients will have to > show this change based off events they receive. You should be able to > see your session as "On the Phone" if you look at the "Session" tab > in the jive messenger admin console. > > You can see the Proto-JEP for more infomation on the packets: > http://svn.jivesoftware.org/svn/repos/asterisk-im/trunk/documentation/ > phone_jep.html > >> >> btw. did you have a look at replacing the jakarta-commons threadpool >> with something else, so we can add that to Asterisk-Java without adding >> a dependency on an external library? > > > Not yet, been working on trying to get the plugin working in some > form. I will try to have this done soon and definitely before the > final release. > > >> >> nice work, >> >> =Stefan >> >> >> >> ------------------------------------------------------- >> SF.Net email is Sponsored by the Better Software Conference & EXPO >> September 19-22, 2005 * San Francisco, CA * Development Lifecycle >> Practices >> Agile & Plan-Driven Development * Managing Projects & Teams * >> Testing & QA >> Security * Process Improvement & Measurement * http://www.sqe.com/ >> bsce5sf >> _______________________________________________ >> Asterisk-java-users mailing list >> Ast...@li... >> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users >> > > > > ------------------------------------------------------- > SF.Net email is Sponsored by the Better Software Conference & EXPO > September 19-22, 2005 * San Francisco, CA * Development Lifecycle > Practices > Agile & Plan-Driven Development * Managing Projects & Teams * Testing > & QA > Security * Process Improvement & Measurement * http://www.sqe.com/bsce5sf > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
From: Andrew W. <an...@ji...> - 2005-08-04 15:41:28
|
Stefan, It is cool that you have had the chance to check this out and it is mostly working for you :) Please see my comments below: On Aug 3, 2005, at 5:23 PM, Stefan Reuter wrote: >> The 1.0 beta1 of the Asterisk-IM Jive Messenger XMPP (jabber) server >> plugin has been released today. >> > > hey, that looks great. > > now we must get the mainstream jabber clients to support the protocol > extension... i just tried it with the xml console of psi and made it > dial. People from Trillian, Gaim, and Pandian have all committed to supporting the plugin. Hopefully we will see something from them in the near future. I am sure others will follow suit afterwards. > > how do i get the events and presence updates? The events should be sent as long as your asterisk manager user has the correct perms. Your manager should have the following privleges: read/write = call,command You won't see the presence change on your client, though other users can see your status changes. This is because Presence packets aren't sent to the user who changed their presence. Clients will have to show this change based off events they receive. You should be able to see your session as "On the Phone" if you look at the "Session" tab in the jive messenger admin console. You can see the Proto-JEP for more infomation on the packets: http://svn.jivesoftware.org/svn/repos/asterisk-im/trunk/documentation/ phone_jep.html > > btw. did you have a look at replacing the jakarta-commons threadpool > with something else, so we can add that to Asterisk-Java without > adding > a dependency on an external library? Not yet, been working on trying to get the plugin working in some form. I will try to have this done soon and definitely before the final release. > > nice work, > > =Stefan > > > > ------------------------------------------------------- > SF.Net email is Sponsored by the Better Software Conference & EXPO > September 19-22, 2005 * San Francisco, CA * Development Lifecycle > Practices > Agile & Plan-Driven Development * Managing Projects & Teams * > Testing & QA > Security * Process Improvement & Measurement * http://www.sqe.com/ > bsce5sf > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > |
From: Stefan R. <sr...@re...> - 2005-08-04 00:23:41
|
> The 1.0 beta1 of the Asterisk-IM Jive Messenger XMPP (jabber) server > plugin has been released today. hey, that looks great. now we must get the mainstream jabber clients to support the protocol extension... i just tried it with the xml console of psi and made it dial. how do i get the events and presence updates? btw. did you have a look at replacing the jakarta-commons threadpool with something else, so we can add that to Asterisk-Java without adding a dependency on an external library? nice work, =Stefan |