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From: Thomas K. <tho...@ac...> - 2008-01-28 09:44:01
|
On Friday 25 January 2008 13:06:38 Stefan Reuter wrote: > Thomas, you said the agent was on a direct call from another agent and > not one from the queue. In Asterisk you have pseudo channels for agents > (Agent/xxx iirc) if you want to treat the calls from other agents like > queue calls you should not call the agent directly (e.g. on the > corresponding SIP/xxx channel but use the agent pseudo channel in your > dialplan. This should make the queue aware that the agent is on call and > trigger the desired events. OK. Thanks a lot for your help. When I try to dial the agent [exten => 1234,n,Dial(Agent/1234)] instead of the SIP channel, then I get the following warning from Asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) I think this problem doesn't really belongs to this mailing-list. But if you have an idea what the problem could be, I would be gratful for any hint. ;-) Regards, Thomas -- Thomas Kenner |
|
From: Breucking P. <bre...@go...> - 2008-01-25 12:13:35
|
Thats my opinion to, it will be difficult to understand processes, if they are handled different in a "live"-image of asterisk Regards, Patrick Patrick Breucking <bre...@GO...> (System Engineer) * GONICUS GmbH * NL Arnsberg * Moehnestrasse 11-17 * D-59755 Arnsberg * Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16 - 278 * http://www.GONICUS.de *Sitz der Gesellschaft: Moehnestrasse 11-17 * D-59755 Arnsberg *Geschaeftsfuehrer: Rainer Luelsdorf, Alfred Schroeder *Vorsitzender des Beirats: Juergen Michels *Amtsgericht Arnsberg * HRB 1968 Am 25.01.2008 um 13:06 schrieb Stefan Reuter: > Breucking Patrick wrote: >> This event is generated by the asterisk and so it should. This is >> maybe >> a fault in the asterisk agent model. Stefan what do you think about >> it. >> Should we "correct" these "faults"? > > I think we should closly mirror the Asterisk model in the live API. > > Thomas, you said the agent was on a direct call from another agent and > not one from the queue. In Asterisk you have pseudo channels for > agents > (Agent/xxx iirc) if you want to treat the calls from other agents like > queue calls you should not call the agent directly (e.g. on the > corresponding SIP/xxx channel but use the agent pseudo channel in your > dialplan. This should make the queue aware that the agent is on call > and > trigger the desired events. > > =Stefan > > -- > reuter network consulting > Neusser Str. 110 > 50760 Koeln > Germany > Telefon: +49 221 1305699-0 > Telefax: +49 221 1305699-90 > E-Mail: ste...@re... > Jabber: ste...@re... > WWW: http://www.reucon.com > > Steuernummern 215/5140/1791 USt-IdNr. DE220701760 > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/_______________________________________________ > Asterisk-java-devel mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-devel |
|
From: Stefan R. <ste...@re...> - 2008-01-25 12:06:57
|
Breucking Patrick wrote: > This event is generated by the asterisk and so it should. This is maybe= > a fault in the asterisk agent model. Stefan what do you think about it.= > Should we "correct" these "faults"? I think we should closly mirror the Asterisk model in the live API. Thomas, you said the agent was on a direct call from another agent and not one from the queue. In Asterisk you have pseudo channels for agents (Agent/xxx iirc) if you want to treat the calls from other agents like queue calls you should not call the agent directly (e.g. on the corresponding SIP/xxx channel but use the agent pseudo channel in your dialplan. This should make the queue aware that the agent is on call and trigger the desired events. =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: ste...@re... Jabber: ste...@re... WWW: http://www.reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 |
|
From: Breucking P. <bre...@go...> - 2008-01-25 11:50:06
|
Am 25.01.2008 um 10:02 schrieb Thomas Kenner: > Hi Patrick, > > The engaged agent answered the call directly from another agent. I > tested > calling over a queue: the state was set to AGENT_ONCALL. Should the > state > only be set to AGENT_ONCALL if the call comes from a queue? This event is generated by the asterisk and so it should. This is maybe a fault in the asterisk agent model. Stefan what do you think about it. Should we "correct" these "faults"? > > > When is the state AGENT_RINGING set? Which event is triggered? This event is not implemented and will be removed. > > > Thanks a lot, > Thomas > > (I'm using code revision 944 and Asterisk 1.4.10) > > > On Thursday 24 January 2008 16:06:55 Breucking Patrick wrote: >> Hi Thomas, >> >> I checked your snippet against trunk version and it worked as >> expected. Has the engaged agent answered a call from the queue? >> >> For more help I need some more information: >> >> - asterisk version >> - code revision >> - short descriptions of agents/queue scenario >> >> Regards, >> Patrick >> >> >> Patrick Breucking <bre...@GO...> (System Engineer) >> * GONICUS GmbH * NL Arnsberg * Moehnestrasse 11-17 * D-59755 Arnsberg >> * Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16 - 278 >> * http://www.GONICUS.de >> >> *Sitz der Gesellschaft: Moehnestrasse 11-17 * D-59755 Arnsberg >> *Geschaeftsfuehrer: Rainer Luelsdorf, Alfred Schroeder >> *Vorsitzender des Beirats: Juergen Michels >> *Amtsgericht Arnsberg * HRB 1968 >> >> Am 24.01.2008 um 15:22 schrieb Thomas Kenner: >>> Hello! >>> >>> I've got the following problem: >>> >>> I'd like to get the number of agents which are in the "calling" >>> state. I wrote >>> the following code: >>> >>> for (AsteriskAgent agent : asteriskServer.getAgents()) { >>> if(agent.getStatus() == AgentState.AGENT_ONCALL){ >>> active_calls++; >>> } >>> } >>> >>> But the state of all agents is always IDLE, even if the >>> corresponding agent is >>> currently engaged in a call. >>> >>> Did I forget to set something in the config files of asterisk? I >>> already set >>> the following variables: >>> in queues.conf: eventwhencalled = yes >>> in sip.conf: callevents = yes >>> >>> Thanks a lot, >>> Thomas >>> >>> -- >>> Thomas Kenner >>> >>> >>> ------------------------------------------------------------------------- >>> This SF.net email is sponsored by: Microsoft >>> Defy all challenges. Microsoft(R) Visual Studio 2008. >>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>> _______________________________________________ >>> Asterisk-java-devel mailing list >>> Ast...@li... >>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-devel > > > > -- > DI Thomas Kenner > eMail: tho...@ac... > www: http://www.acoveo.com > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Asterisk-java-devel mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-devel > |
|
From: Thomas K. <tho...@ac...> - 2008-01-25 09:02:35
|
Hi Patrick, The engaged agent answered the call directly from another agent. I tested calling over a queue: the state was set to AGENT_ONCALL. Should the state only be set to AGENT_ONCALL if the call comes from a queue? When is the state AGENT_RINGING set? Which event is triggered? Thanks a lot, Thomas (I'm using code revision 944 and Asterisk 1.4.10) On Thursday 24 January 2008 16:06:55 Breucking Patrick wrote: > Hi Thomas, > > I checked your snippet against trunk version and it worked as > expected. Has the engaged agent answered a call from the queue? > > For more help I need some more information: > > - asterisk version > - code revision > - short descriptions of agents/queue scenario > > Regards, > Patrick > > > Patrick Breucking <bre...@GO...> (System Engineer) > * GONICUS GmbH * NL Arnsberg * Moehnestrasse 11-17 * D-59755 Arnsberg > * Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16 - 278 > * http://www.GONICUS.de > > *Sitz der Gesellschaft: Moehnestrasse 11-17 * D-59755 Arnsberg > *Geschaeftsfuehrer: Rainer Luelsdorf, Alfred Schroeder > *Vorsitzender des Beirats: Juergen Michels > *Amtsgericht Arnsberg * HRB 1968 > > Am 24.01.2008 um 15:22 schrieb Thomas Kenner: > > Hello! > > > > I've got the following problem: > > > > I'd like to get the number of agents which are in the "calling" > > state. I wrote > > the following code: > > > > for (AsteriskAgent agent : asteriskServer.getAgents()) { > > if(agent.getStatus() == AgentState.AGENT_ONCALL){ > > active_calls++; > > } > > } > > > > But the state of all agents is always IDLE, even if the > > corresponding agent is > > currently engaged in a call. > > > > Did I forget to set something in the config files of asterisk? I > > already set > > the following variables: > > in queues.conf: eventwhencalled = yes > > in sip.conf: callevents = yes > > > > Thanks a lot, > > Thomas > > > > -- > > Thomas Kenner > > > > > > ------------------------------------------------------------------------- > > This SF.net email is sponsored by: Microsoft > > Defy all challenges. Microsoft(R) Visual Studio 2008. > > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > > _______________________________________________ > > Asterisk-java-devel mailing list > > Ast...@li... > > https://lists.sourceforge.net/lists/listinfo/asterisk-java-devel -- DI Thomas Kenner eMail: tho...@ac... www: http://www.acoveo.com |
|
From: Breucking P. <bre...@go...> - 2008-01-24 15:07:00
|
Hi Thomas, I checked your snippet against trunk version and it worked as expected. Has the engaged agent answered a call from the queue? For more help I need some more information: - asterisk version - code revision - short descriptions of agents/queue scenario Regards, Patrick Patrick Breucking <bre...@GO...> (System Engineer) * GONICUS GmbH * NL Arnsberg * Moehnestrasse 11-17 * D-59755 Arnsberg * Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16 - 278 * http://www.GONICUS.de *Sitz der Gesellschaft: Moehnestrasse 11-17 * D-59755 Arnsberg *Geschaeftsfuehrer: Rainer Luelsdorf, Alfred Schroeder *Vorsitzender des Beirats: Juergen Michels *Amtsgericht Arnsberg * HRB 1968 Am 24.01.2008 um 15:22 schrieb Thomas Kenner: > Hello! > > I've got the following problem: > > I'd like to get the number of agents which are in the "calling" > state. I wrote > the following code: > > for (AsteriskAgent agent : asteriskServer.getAgents()) { > if(agent.getStatus() == AgentState.AGENT_ONCALL){ > active_calls++; > } > } > > But the state of all agents is always IDLE, even if the > corresponding agent is > currently engaged in a call. > > Did I forget to set something in the config files of asterisk? I > already set > the following variables: > in queues.conf: eventwhencalled = yes > in sip.conf: callevents = yes > > Thanks a lot, > Thomas > > -- > Thomas Kenner > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Asterisk-java-devel mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-devel > |
|
From: Carlos G M. <tr...@hu...> - 2008-01-24 14:50:38
|
Thomas,
if you are using Agents, they are only "ACTIVE" when receiving calls.
If they make calls, then the Agent channel is still idle, as you see.
Managing this and also the fact that most IP phones can actually
have multiple calls is ... problematic sometimes.
(I.e. your agents, if human, usually don't want to receive
an ACD call when talking, and somehow if you have blended
agents, you have to block that).
-Carlos
Thomas Kenner @ 24/1/2008 08:22 -0600 dixit:
> Hello!
>
> I've got the following problem:
>
> I'd like to get the number of agents which are in the "calling" state. I wrote
> the following code:
>
> for (AsteriskAgent agent : asteriskServer.getAgents()) {
> if(agent.getStatus() == AgentState.AGENT_ONCALL){
> active_calls++;
> }
> }
>
> But the state of all agents is always IDLE, even if the corresponding agent is
> currently engaged in a call.
>
> Did I forget to set something in the config files of asterisk? I already set
> the following variables:
> in queues.conf: eventwhencalled = yes
> in sip.conf: callevents = yes
>
> Thanks a lot,
> Thomas
>
--
Carlos G Mendioroz <tr...@hu...> LW7 EQI Argentina
|
|
From: Thomas K. <tho...@ac...> - 2008-01-24 14:22:15
|
Hello!
I've got the following problem:
I'd like to get the number of agents which are in the "calling" state. I wrote
the following code:
for (AsteriskAgent agent : asteriskServer.getAgents()) {
if(agent.getStatus() == AgentState.AGENT_ONCALL){
active_calls++;
}
}
But the state of all agents is always IDLE, even if the corresponding agent is
currently engaged in a call.
Did I forget to set something in the config files of asterisk? I already set
the following variables:
in queues.conf: eventwhencalled = yes
in sip.conf: callevents = yes
Thanks a lot,
Thomas
--
Thomas Kenner
|
|
From: Stefan R. <ste...@re...> - 2008-01-19 07:51:04
|
Jesus Mogollon wrote: > to properly create the string. How could I send it over for review? Please open an issue at jira.reucon.org and attach the patch (preferred against latest svn trunk) Thanks, Stefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: ste...@re... Jabber: ste...@re... WWW: http://www.reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 |
|
From: Jesus M. <jmo...@gm...> - 2008-01-19 03:26:01
|
Greetings all
Has anybody tested UpdateConfigAction? I'm in the middle of a project
and, when I tried to use it, it didn't work. I found a couple of problems:
1) The numbers following Action-XXXXXX, Cat-XXXXX and so on are not
zero-padded in addCommand.
2) When the map created in addCommand is processed by ActionBuilderImpl, it
creates the lines with "action: Cat-000000=<STRING>" when the value should
be
"Cat-000000: <STRING>". Note the space after the ":". It would not work
without it.
I think I fixed most of the problems already by zero-padding the number
before append it to the command string and by checking if the action is an
instance of UpdateConfig in ActionBuilderImpl.java in order to properly
create the string. How could I send it over for review?
Jesus Mogollon
|
|
From: Stefan R. <ste...@re...> - 2007-12-19 16:17:53
|
Martin Smith wrote: > I should add that you could get Voicemail recordings if you used a > backend that supported pulling the recordings out, such as IMAP. Yes that would be an option. Another one would be to run a small Webserver on the Asterisk box that makes the voicemail files available through HTTP. =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: ste...@re... Jabber: ste...@re... Steuernummern 215/5140/1791 USt-IdNr. DE220701760 |
|
From: Martin S. <ma...@be...> - 2007-12-19 15:10:36
|
I should add that you could get Voicemail recordings if you used a backend that supported pulling the recordings out, such as IMAP. =20 Martin Smith, Systems Developer ma...@be... Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221=20 =20 ________________________________ From: ast...@li... [mailto:ast...@li...] On Behalf Of Juan Freitas Sent: Wednesday, December 19, 2007 7:26 AM To: ast...@li... Subject: [Asterisk-java-devel] Getting voicemail from a mailbox =09 =09 Hello everyone, =20 I'm a newbie, and I was wondering if there is any way to retrieve de message left in someone voicemailbox using the asterisk-java API. If so, can you give some hints or examples of how to do this? =20 =20 Many thanks, =20 Juan Freitas =09 =09 =20 =20 |
|
From: Martin S. <ma...@be...> - 2007-12-19 15:07:57
|
Hi Juan, =20 If you mean retrieve the recording as a file, then no. You'll need to get on the machine running Asterisk to get at the file. If you mean leave or play voicemail over AGI, then yes, just run the Voicemail(Main) application(s). With the Manager API, you could also send calls outward that played voicemail. =20 If you're looking for file transfer, then no, Asterisk's Manager API and Gateway Interface both don't support that. =20 =20 Martin Smith, Systems Developer ma...@be... Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221=20 =20 ________________________________ From: ast...@li... [mailto:ast...@li...] On Behalf Of Juan Freitas Sent: Wednesday, December 19, 2007 7:26 AM To: ast...@li... Subject: [Asterisk-java-devel] Getting voicemail from a mailbox =09 =09 Hello everyone, =20 I'm a newbie, and I was wondering if there is any way to retrieve de message left in someone voicemailbox using the asterisk-java API. If so, can you give some hints or examples of how to do this? =20 =20 Many thanks, =20 Juan Freitas =09 =09 =20 =20 |
|
From: Juan F. <jua...@no...> - 2007-12-19 12:25:50
|
Hello everyone, =20 I'm a newbie, and I was wondering if there is any way to retrieve de message left in someone voicemailbox using the asterisk-java API. If so, can you give some hints or examples of how to do this? =20 =20 Many thanks, =20 Juan Freitas =20 =20 |
|
From: lumen <lu...@bu...> - 2007-12-17 08:16:31
|
Hi all:
I find a bug in org.asteriskjava.fastagi.DefaultAgiServer in loadConfig()
method, using 0.3.1 of asterisk-java.
The problem is that for backward compatibylity, firts it makes a lookup of
'port' resource, and if that returns an emtpy string, it makes a lookup of
'bindPort' resource.
The problem is that if 'port' is not found an exception raises, and the
second lookup is never entered.
The 'if' block:
if (portString == null)
{
// for backward compatibility only
portString = resourceBundle.getString("bindPort");
}
have to be moved outside the try{}catch.
How I can report this bug at http://jira.reucon.org/browse/AJ ? I can't
find the form to report a new bug with the patch.
|
|
From: Martin S. <ma...@be...> - 2007-11-28 17:41:14
|
FYI -- changes incoming to trunk.=20 Martin Smith, Systems Developer ma...@be... Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221=20 -----Original Message----- From: ast...@li... [mailto:ast...@li...] On Behalf Of Johansson Olle E Sent: Wednesday, November 28, 2007 11:52 AM To: Asterisk Mailing List Developers Subject: [asterisk-dev] AMI update in SVN trunk based on "moremanager"branch * Changes to the AMI API! Friends, A while ago I created a branch based on 1.4 called "moremanager" to =20 enhance the manager interface. During that work we agreed on the changes that are now being =20 integrated into trunk. This will mean that trunk will start working differently for those of you using manager. I've documented all the changes in a README, documentation that will =20 be integrated too. http://svn.digium.com/view/asterisk/team/oej/moremanager/CHANGES.moreman ager?view=3Dmarkup I encourage all AMI client developers to start using the =20 "CoreSettings" command after connection. If it doesn't work, your Asterisk is using some version of AMI 1.0. If it does work, you get a report of Asterisk Version and AMI version. =20 Whenever we change AMI in the future, we will change the AMI version number so you can keep =20 track of what you're in communication with. There are a lot of cleanups and changes to existing actions, some new =20 actions and some new events. You can now also unload, load and reload modules by using the AMI. =20 When a module is properly loaded and configured, you will get a report so you can start =20 interacting with it. I will continue merging these changes over the next week or so. Any =20 feedback is, as always, appreciated. Cheers from a dark and cold Sweden! /Olle _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev |
|
From: <ash...@wi...> - 2007-10-31 10:05:42
|
Hi all,
=20
I want to make a SIP soft phone in Java using Asterisk as
SIP server.
=20
Is there any library available for this?
=20
Asterisk-java is there, but it does not give solution for
answering call.
=20
Can anybody tell me any SIP Java library which is compatible
with Asterisk apart from Asterisk-Java?
=20
=20
=20
=20
Regards,
Ashish
=20
=20
|
|
From: Martin S. <ma...@be...> - 2007-10-26 13:44:45
|
Hello Ashish, =20 First off, the Asterisk Manager interface isn't going to provide you enough functionality to make a softphone. It does NOT handle the media path at all, so you'll never have sound or video or anything else via the Manager interface. You're going to need to use SIP or IAX (and you'll find Java libraries for each, though I'd argue they might be a bit immature still). Given all that, Asterisk-Java's manager interface might add some advanced features to a pre-existing softphone. =20 Regarding answer a call, you cannot force a remote endpoint to answer a call; you also cannot be a remote-end capable of receiving calls if you are only using Asterisk-Java's manager interface. If you have some other remote end, you can certain use the manager's Originate and Bridge actions to get channels connected to each other. You can also use local channels there, which should provide you a lot of flexibility. =20 Hope this helps, =20 Martin Smith, Systems Developer ma...@be... Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221=20 =20 ________________________________ From: ast...@li... [mailto:ast...@li...] On Behalf Of ash...@wi... Sent: Friday, October 26, 2007 5:44 AM To: ast...@li... Subject: [Asterisk-java-devel] Answering call using Asterisk Java. =09 =09 Hi Friends, =20 I am developing a soft phone using Asterisk Java API and asterisk as server. =20 I am using Manager API. I am able to call another extension but unable receive the call From another extension. I can not find the method in Manager API. =20 Can anybody tell me how I can answer the call using Manager API? =20 =20 Thanks!! Regards, Ashish |
|
From: <ash...@wi...> - 2007-10-26 10:01:45
|
Hi all,
=20
I am Ashish here.
I am using Manager API for developing a softphone.
=20
I can dial to extension using OriginateAction.
But unable to answer call coming to that extension
As I can not find appropriate method for same.
=20
Anyone tell me how can answer call using Manager API?
=20
=20
Regards,
Ashish
=20
=20
=20
|
|
From: <ash...@wi...> - 2007-10-26 09:44:45
|
Hi Friends,
=20
I am developing a soft phone using Asterisk Java API and
asterisk as server.
=20
I am using Manager API. I am able to call another extension
but unable receive the call
>From another extension. I can not find the method in Manager API.
=20
Can anybody tell me how I can answer the call using Manager
API?
=20
=20
Thanks!!
Regards,
Ashish
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From: Stefan R. <ste...@re...> - 2007-10-22 15:42:18
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hoaianh ngovi wrote: > I have a lot of AGI-stuffs that needed database support. In order to > enhance perfomance I plan to extend DefaultAgiServer to hold a > (database) Connection Pool. Has someone already done that? There is no need to extend the AgiServer in that case. By default there is only one instance of an agi script so you can just create a common base class for you agi scripts and hold the database pool there. =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: ste...@re... Jabber: ste...@re... Steuernummern 215/5140/1791 USt-IdNr. DE220701760 |
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From: hoaianh n. <ho...@gm...> - 2007-10-22 12:25:17
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Hi dear developers, I have a lot of AGI-stuffs that needed database support. In order to enhance perfomance I plan to extend DefaultAgiServer to hold a (database) Connection Pool. Has someone already done that? How can I access the objects holded within the AGI Server? Thank you and regards -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kanns mit allen: http://www.gmx.net/de/go/multimessenger |
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From: Patrick B. <bre...@go...> - 2007-09-14 12:43:24
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Hi developers, I thought about a nice new feature of live API: An event for QueueListener which will be generated, if a call (channel) reaches the queue's service level (and maybe the timeout, a optional setting in queues.conf). How do you think about it? Here is my implementation approach: 0. Queue has a private Map with <Channel, Timer> 1a. When a channel enters a queue: Add a Timer (key is channel) to Map. TimerTask notifies Queue if SL exceeded. Delay of Timer is the queue service level. 1b. Server notify all Listeners and send the channel wich exceeded. 2. When channel leaves the queue, stop timer and remove it Regards Patrick -- Patrick Breucking <bre...@GO...> (System Engineer) * GONICUS GmbH * NL Arnsberg * Moehnestrasse 11-17 * D-59755 Arnsberg * Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16 - 278 * http://www.GONICUS.de *Sitz der Gesellschaft: Moehnestrasse 11-17 * D-59755 Arnsberg *Geschaeftsfuehrer: Rainer Luelsdorf, Alfred Schroeder *Vorsitzender des Beirats: Juergen Michels *Amtsgericht Arnsberg * HRB 1968 |
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From: Stefan R. <ste...@re...> - 2007-09-12 21:03:56
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Hi, I have configured the FishEye-Jira integration for Asterisk-Java. This means you will see your changes in Jira associated with the correct issue. To make this work make sure you include the issue id in your commit message like [AJ-XX] Fix for blah You can see an example at http://jira.reucon.org/browse/AJ-88?page=3Dcom.cenqua.fisheye.jira:fishey= e-issuepanel (select the FishEye tab on any issue) =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: ste...@re... Jabber: ste...@re... Steuernummern 215/5140/1791 USt-IdNr. DE220701760 |
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From: Patrick B. <bre...@go...> - 2007-09-12 13:10:09
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Hi coders, today I checked in my code for Asterisk agents management inside the live API. I would add support for QueueMembers in AsteriskQueues. QueueMembers can be Agents or SIP-Accounts. To build this link inside the Live API I would do the following: 1. Add a Property (Collection) to AsteriskAgent called QueueMembership. 2. On QueueManager.initialize react on QueueMemberEvents and create QueueMembers (new class). On creation try to retrieve a representing agent and add the queue to agent (as Membership) (3. Maybe it will be usefull to store the instance of the representing agent in the QueueMember) What you think about? Regards Patrick Breucking -- Patrick Breucking <bre...@GO...> (System Engineer) * GONICUS GmbH * NL Arnsberg * Moehnestrasse 11-17 * D-59755 Arnsberg * Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16 - 278 * http://www.GONICUS.de *Sitz der Gesellschaft: Moehnestrasse 11-17 * D-59755 Arnsberg *Geschaeftsfuehrer: Rainer Luelsdorf, Alfred Schroeder *Vorsitzender des Beirats: Juergen Michels *Amtsgericht Arnsberg * HRB 1968 -- Patrick Breucking <bre...@GO...> (System Engineer) * GONICUS GmbH * NL Arnsberg * Moehnestrasse 11-17 * D-59755 Arnsberg * Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16 - 278 * http://www.GONICUS.de *Sitz der Gesellschaft: Moehnestrasse 11-17 * D-59755 Arnsberg *Geschaeftsfuehrer: Rainer Luelsdorf, Alfred Schroeder *Vorsitzender des Beirats: Juergen Michels *Amtsgericht Arnsberg * HRB 1968 |
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