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From: Stefan R. <ste...@re...> - 2007-03-08 20:52:09
|
S. Brett Sutton wrote: > It would still be nice to suppress the warning if either of the two > prompts came back. Sure, I've added it to svn. > In the mean time I will encourage our dev team to upgrade to a later > version. >=20 > I assume that m2 is fairly stable? Stable in the sense of "works without any known bugs" yes, stable in the sense of feature complete and won't change no. =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: S. B. S. <bs...@no...> - 2007-03-08 20:00:14
|
I will have to check. But it will be one of the older releases. It would still be nice to suppress the warning if either of the two prompts came back. In the mean time I will encourage our dev team to upgrade to a later version. I assume that m2 is fairly stable? Stefan Reuter wrote: > S. Brett Sutton wrote: > >> Stephen, >> we have a user trying to use OpenPBX when they discovered the >> following problem: >> >> /"Ok, I figured it out. >> It indeed was that AsterFax was looking for Asterisk Call Manager/1.0 >> instead of OpenPBX Call Manager/1.0 >> >> Stupid as that. >> >> I just went into the OpenPBX source code and changed what Openpbx >> answers when connecting to the API."/ >> >> Could you change the asterisk-java code base so that it accepts both >> '/Asterisk Call Manager/1.0' and '//OpenPBX Call Manager/1.0'. >> > > Which version of Asterisk-Java do you use? > At least with current svn and 0.3-m2 you should only get a warning if > the exact identifier isn't recognized. > > =Stefan > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > |
From: Stefan R. <ste...@re...> - 2007-03-08 19:48:17
|
S. Brett Sutton wrote: > Stephen, > we have a user trying to use OpenPBX when they discovered the > following problem: >=20 > /"Ok, I figured it out. > It indeed was that AsterFax was looking for Asterisk Call Manager/1.0 > instead of OpenPBX Call Manager/1.0 >=20 > Stupid as that. >=20 > I just went into the OpenPBX source code and changed what Openpbx > answers when connecting to the API."/ >=20 > Could you change the asterisk-java code base so that it accepts both > '/Asterisk Call Manager/1.0' and '//OpenPBX Call Manager/1.0'. Which version of Asterisk-Java do you use? At least with current svn and 0.3-m2 you should only get a warning if the exact identifier isn't recognized. =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: S. B. S. <bs...@no...> - 2007-03-08 19:30:22
|
Stephen, we have a user trying to use OpenPBX when they discovered the following problem: /"Ok, I figured it out. It indeed was that AsterFax was looking for Asterisk Call Manager/1.0 instead of OpenPBX Call Manager/1.0 Stupid as that. I just went into the OpenPBX source code and changed what Openpbx answers when connecting to the API."/ Could you change the asterisk-java code base so that it accepts both '/Asterisk Call Manager/1.0' and '//OpenPBX Call Manager/1.0'. /Regards, Brett/ / Stefan Reuter wrote: > Henrik Buchholz wrote: > >> [INFO] Cannot execute mojo: resources. It requires a project with an >> existing pom.xml, but the build is not using one. >> > > You must call mvn install in the same directory that contains the > pom.xml. Did you do that? > > =Stefan > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > |
From: Stefan R. <ste...@re...> - 2007-03-08 19:14:23
|
Henrik Buchholz wrote: > [INFO] Cannot execute mojo: resources. It requires a project with an > existing pom.xml, but the build is not using one. You must call mvn install in the same directory that contains the pom.xml. Did you do that? =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: Henrik B. <hen...@fr...> - 2007-03-08 14:50:38
|
Hi, I'm trying to build the asterisk-java jar from the svn source: mvn install always throws the following: [INFO] Scanning for projects... [INFO] ---------------------------------------------------------------------------- [INFO] Building Maven Default Project [INFO] task-segment: [install] [INFO] ---------------------------------------------------------------------------- [INFO] ------------------------------------------------------------------------ [ERROR] BUILD ERROR [INFO] ------------------------------------------------------------------------ [INFO] Cannot execute mojo: resources. It requires a project with an existing pom.xml, but the build is not using one. [INFO] ------------------------------------------------------------------------ [INFO] For more information, run Maven with the -e switch [INFO] ------------------------------------------------------------------------ [INFO] Total time: 1 second [INFO] Finished at: Thu Mar 08 15:48:36 CET 2007 [INFO] Final Memory: 3M/5M [INFO] ------------------------------------------------------------------------ of course the pom.xml sits in the directory. Am I missing something? thanks Henrik |
From: Dinesh S. <din...@gm...> - 2007-03-07 14:03:21
|
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <title>Dinesh has Tagged you! :)</title> <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> </head> <body> <div style="padding: 0 0 10px 30px"><a href="http://www.taggedmail.com/welcome.html?conn=1trs3oiw0&ect=3k1qt17&tId=130065&fid=3e369a88fafe6010"><img src="http://www.taggedmail.com/imgld.php?img=logo_email_small.gif&ect=3k1qt17" width="98" height="35" alt="" border="0"></a></div> <table border="0" cellspacing="0" cellpadding="0" width="539"> <tr> <td colspan="3"><img src="http://static.tagged.com/images/invite_crnrrs_topLong.gif" width="539" height="11" alt="" border="0"></td> </tr> <tr> <td align="left" bgcolor="#c9c9c9" width="1"></td> <td width="537" background="http://static.tagged.com/images/invite_gradient.gif" style="font-size: 30px; color: #000000; text-align: center; font-family: Arial, Helvetica, sans-serif; font-weight: bold; background-image: url(http://static.tagged.com/images/invite_gradient.gif); background-repeat: repeat-x; height: 194px;"> <div style="float: left; padding: 15px 10px 0 10px; text-align:center; width:110px; font-size: 13px;"> <img src="http://www.taggedmail.com/imgsrv.php?uid=21040245" /><br /> <div style="padding:5px 0 0 0;">Dinesh S, 22</div> <div style="font-size: 12px; color: red;">Minot, ME</div> </div> <div style="float: left; width:405px;"> <div style="font-size:16px; padding:0 0 10px 0;">Dinesh S has added you as a friend</div> <div style="font-size:20px; margin:0; padding:0 0 15px 0;">Is Dinesh S your friend?</div> <a href="http://www.taggedmail.com/welcome.html?conn=1trs3oiw0&ect=3k1qt17&tId=130065&fid=3e369a88fafe6010" style="text-decoration:none;"><img src="http://static.tagged.com/images/btn_email_yes.gif" width="83" height="37" alt="" border="0" /> <img src="http://static.tagged.com/images/btn_email_no.gif" width="83" height="37" alt="" border="0" /></a> <div style="font-size:16px; padding:20px 0 0 0;">Please respond or Dinesh may think you said no :(</div> </div> </td> <td align="right" bgcolor="#c9c9c9" width="1"></td> </tr> <tr> <td colspan="3"><img src="http://static.tagged.com/images/invite_crnrs_bottomShort.gif" width="539" height="11" alt="" border="0"></td> </tr> <tr> <td colspan="3"> <div style="font-size: 12px; font-color: black; font-family: Arial, Helvetica, sans-serif; padding: 10px 0 0 0; text-align: center;"> <a style="color: black" href="http://www.taggedmail.com/no_more.html?unsem=asterisk-java-users%40lists.sourceforge.net&tId=130065&fid=3e369a88fafe6010">Click here</a> to unsubscribe from Tagged, P.O. Box 193152 San Francisco, CA 94119-3152 </div> </td> </tr> </table> </body> </html> <!-- k2ngu --> |
From: Gaetan M. <gm...@ea...> - 2007-02-16 10:41:18
|
Hi Stefan If you're interested here is a small patch against db actions and event: - Fix DBGetREsponseEvent, the key for the value is "value" and no "val" (at least in my * 1.2.9) , so added an additional setValue() on the event. - Fix DBPutAction also, key is value and not val (see comment about documentation mistakes on voip-info.org's page for dbput). - Created DBDelAction (quite recent addition to the manager api in spring 2006 ?). Gaetan |
From: Gaetan M. <gm...@ea...> - 2007-02-16 09:54:22
|
Hi, I'm using asterisk-live and the OriginateCallback interface to track originate requests. >From time to time, I get an error in my code because it seems onDialing() is not always called. Indeed I can detect in the onSuccess callback that onDialing has never been called previously. Are there known situations where this can happen ? I have not been able yet to tell if it's the asterisk manager that skips events or asterisk-live that misses some ? This happens when I originate from an sip channel that's in autoanswer mode (operator with headset) so maybe something is going too fast for the manager or asterisk-java ? regards, Gaetan |
From: Stefan R. <ste...@re...> - 2007-02-16 04:53:34
|
Burak Do=C4=9Fru=C3=B6z wrote: > I was just curious about my hangup routine being improper. It seems to be ok. However I am not sure about hanging up both the source and destination port. If these two channels are linked it is sufficient to hangup one of them as the other is hung up automatically in that case. > Anyway i get > a timeout exception if i set the timeout value to 50000. Is this a smal= l > value or am i doing something wrong while getting a response? Is it ok > to leave it 1 so my method can send consecutive hangup actions. (In > practice i don=E2=80=99t need hangup results) 50000 is rather large for a HangupAction. If you are not interested in the response you should pass null instead - that turns the hangup action into an asynchronous message and returns immediately. =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... |
From: <bdo...@ip...> - 2007-02-15 09:19:28
|
This happens when i do a quick series of hangup-redials so I cannot test it directly from cli or by telnetting to manager api. However i found out that this happens when i call a spesific model SIP phone. Testing it with another model resulted in a different behaviour (it contined to ring after hangup but returned to normal after picking up the dummy ringing call) So this problem should be set model or asterisk related (As the prior one locked and didn't return to normal until calling reload command on asterisk). I was just curious about my hangup routine being improper. Anyway i get a timeout exception if i set the timeout value to 50000. Is this a small value or am i doing something wrong while getting a response? Is it ok to leave it 1 so my method can send consecutive hangup actions. (In practice i don't need hangup results) Thanks for your concern Burak Re: Consecutive Hangup Action Locking SIP Phone From: Stefan Reuter <stefan.reuter@re...> - 2007-02-14 14:31 You could try to use the Asterisk CLI and do a "soft hangup <channel>" there to verify if this problem is specific to your Java application or a general Asterisk problem. I've never encountered a problem like this and i am hanging up quite a lot channels through Asterisk-Java :) =Stefan -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. |
From: Stefan R. <ste...@re...> - 2007-02-14 23:36:15
|
Gaetan Minet wrote: > Here are some modification you might maybe find usefull ? Patch is > against trunk. Thanks for your patch. I've applied it to trunk. =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... |
From: Stefan R. <ste...@re...> - 2007-02-14 22:31:15
|
Burak Do=F0ru=F6z wrote: > i want to block calls by using some program logic so i wrote a program > to catch dial events, parse them and hangup both destination and source= > channels if they fit in my blocking logic. >=20 > However, if i hang up consecutive calls for a particular sip phone, the= > sip phone gets locked and it becomes unreachable and gives busy signal > (cannot be called and it cannot call, just busy) This cannot be fixed b= y > resetting phones but only calling reload command on asterisk. You could try to use the Asterisk CLI and do a "soft hangup <channel>" there to verify if this problem is specific to your Java application or a general Asterisk problem. I've never encountered a problem like this and i am hanging up quite a lot channels through Asterisk-Java :) =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... |
From: <bdo...@ip...> - 2007-02-13 09:47:43
|
Hello, i want to block calls by using some program logic so i wrote a program to catch dial events, parse them and hangup both destination and source channels if they fit in my blocking logic. However, if i hang up consecutive calls for a particular sip phone, the sip phone gets locked and it becomes unreachable and gives busy signal (cannot be called and it cannot call, just busy) This cannot be fixed by resetting phones but only calling reload command on asterisk. My asterisk is 1.2.12.1 and asterisk-java is 0.3-m2 My Hangup routine is like this: public int hangup(String sport,String dport) { try { this.managerConnection.sendAction(new HangupAction(sport),1); } catch(Exception e) {} try { this.managerConnection.sendAction(new HangupAction(dport),1); } catch(Exception e) {} return 0; } What may be the cause and how to fix this? Thanks for your concern Burak -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. |
From: Stefan R. <ste...@re...> - 2007-02-12 10:12:09
|
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, we've just released Asterisk-Java 0.3-m2 at http://asterisk-java.org. The Asterisk-Java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager API. Asterisk-Java is free software distributed under the terms of the Apache License 2.0. Here is the Changelog: Bug * [AJ-47] - AGI does not support multi line data * [AJ-51] - Problems with non-english locales * [AJ-52] - Fix shutdown when using the live api Improvement * [AJ-41] - Add ability to get ManagerConnection from AsteriskServer * [AJ-49] - Support socket read timeout New Feature * [AJ-35] - Support timestamp property on manager events * [AJ-42] - Add support QueueSummary action to Queue manager interface * [AJ-44] - Support PauseMonitor and UnpauseMonitor actions * [AJ-45] - Support ZapRestart action Task * [AJ-53] - Refactor BaseAgiScript to extend AgiOperations and implement AgiScript so users can extend AgiOperation to provide their own add-on features Have fun, Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF0D1pcVCZDfrn+pMRAv9GAKCB1Ozij3l5eLGOu1pOx7kq+PHG4wCePsjR Jm+tu6reyyrDtVDGgATnTwI= =My0x -----END PGP SIGNATURE----- |
From: Mattias M. <ma...@wx...> - 2007-02-08 13:08:19
|
Ok...A quick dump showed that Asterisk 1.2.14 doesn't send out any new = events. The only difference is simply that the ExtensionStatusEvent is = missing... Very strange... Mattias Malmquist - Engineering, wx3 Telecom AB=20 -----Ursprungligt meddelande----- Fr=E5n: ast...@li... = [mailto:ast...@li...] F=F6r Mattias = Malmquist Skickat: den 7 februari 2007 13:26 Till: ast...@li... =C4mne: Re: [Asterisk-java-users] Missing ExtensionStatusEvent I haven't tried sniffing the net, but I logged all events recieved by my = Asterisk-Java application and the ExtensionStateEvent was simply not = there any more. We have downgraded our PBX until we find a reason and a = solution for this. I will consider trying a TCP-dump on a test machine however. Thank you = for the advice. I will post any results here... Best Regards Mattias Malmquist - Engineering, wx3 Telecom AB=20 -----Ursprungligt meddelande----- Fr=E5n: ast...@li... = [mailto:ast...@li...] F=F6r Stefan = Reuter Skickat: den 7 februari 2007 12:17 Till: ast...@li... =C4mne: Re: [Asterisk-java-users] Missing ExtensionStatusEvent -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I am still with 1.2.13, but can try 1.2.14 later if it helps. What you can always do is sniffing the net to see what events Asterisk sends to determine whether its a problem that can be fixed in Asterisk-Java (event syntax changed and event isnt recognized) or Asterisk (event isnt sent at all). =3DStefan Mattias Malmquist wrote: > Hi, >=20 > Since upgrading from Asterisk 1.2.7 to 1.2.14 I suddenly don't recieve > the ExtensionStatusEvent any longer. I have tried version 1.2.13 and = did > receive them there. Do any of you know of any changes between these > versions that may trigger this behaviour? Do any of you have a = solution > for the problem? >=20 > I know this is not an Asterisk-Java specific question, but I figured = it > might be of interest to someone else at this forum... >=20 > Best regards >=20 > Mattias Malmquist - Engineering, wx3 Telecom AB=20 >=20 >=20 >=20 > = -------------------------------------------------------------------------= > Using Tomcat but need to do more? Need to support web services, = security? > Get stuff done quickly with pre-integrated technology to make your job = easier. > Download IBM WebSphere Application Server v.1.0.1 based on Apache = Geronimo > = http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat=3D= 121642 > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFybUlcVCZDfrn+pMRAhP6AJ4/cNkoTy8LDVRBFQXF/XoHph8CyACcCByo V9VlCuUw5iEWFJJGmXplJd4=3D =3D49fB -----END PGP SIGNATURE----- -------------------------------------------------------------------------= Using Tomcat but need to do more? Need to support web services, = security? Get stuff done quickly with pre-integrated technology to make your job = easier. Download IBM WebSphere Application Server v.1.0.1 based on Apache = Geronimo http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat=3D= 121642 _______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users -------------------------------------------------------------------------= Using Tomcat but need to do more? Need to support web services, = security? Get stuff done quickly with pre-integrated technology to make your job = easier. Download IBM WebSphere Application Server v.1.0.1 based on Apache = Geronimo http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat=3D= 121642 _______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
From: dinesh s. <din...@gm...> - 2007-02-08 06:39:15
|
hi , I have been working in asterisk server using java API's. and I need to know the basic connention of asterisk server like how the call handled in asterisk server ????? how the events are occured in asterisk server and how it is estsblished????? becoz i need to build the API's using another language(erlang).so i need some of the basic things. i have got your mailid from search.... Thanks & regards Dinesh.K |
From: Mattias M. <ma...@wx...> - 2007-02-07 12:31:51
|
I haven't tried sniffing the net, but I logged all events recieved by my = Asterisk-Java application and the ExtensionStateEvent was simply not = there any more. We have downgraded our PBX until we find a reason and a = solution for this. I will consider trying a TCP-dump on a test machine however. Thank you = for the advice. I will post any results here... Best Regards Mattias Malmquist - Engineering, wx3 Telecom AB=20 -----Ursprungligt meddelande----- Fr=E5n: ast...@li... = [mailto:ast...@li...] F=F6r Stefan = Reuter Skickat: den 7 februari 2007 12:17 Till: ast...@li... =C4mne: Re: [Asterisk-java-users] Missing ExtensionStatusEvent -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I am still with 1.2.13, but can try 1.2.14 later if it helps. What you can always do is sniffing the net to see what events Asterisk sends to determine whether its a problem that can be fixed in Asterisk-Java (event syntax changed and event isnt recognized) or Asterisk (event isnt sent at all). =3DStefan Mattias Malmquist wrote: > Hi, >=20 > Since upgrading from Asterisk 1.2.7 to 1.2.14 I suddenly don't recieve > the ExtensionStatusEvent any longer. I have tried version 1.2.13 and = did > receive them there. Do any of you know of any changes between these > versions that may trigger this behaviour? Do any of you have a = solution > for the problem? >=20 > I know this is not an Asterisk-Java specific question, but I figured = it > might be of interest to someone else at this forum... >=20 > Best regards >=20 > Mattias Malmquist - Engineering, wx3 Telecom AB=20 >=20 >=20 >=20 > = -------------------------------------------------------------------------= > Using Tomcat but need to do more? Need to support web services, = security? > Get stuff done quickly with pre-integrated technology to make your job = easier. > Download IBM WebSphere Application Server v.1.0.1 based on Apache = Geronimo > = http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat=3D= 121642 > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFybUlcVCZDfrn+pMRAhP6AJ4/cNkoTy8LDVRBFQXF/XoHph8CyACcCByo V9VlCuUw5iEWFJJGmXplJd4=3D =3D49fB -----END PGP SIGNATURE----- -------------------------------------------------------------------------= Using Tomcat but need to do more? Need to support web services, = security? Get stuff done quickly with pre-integrated technology to make your job = easier. Download IBM WebSphere Application Server v.1.0.1 based on Apache = Geronimo http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat=3D= 121642 _______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
From: Stefan R. <ste...@re...> - 2007-02-07 11:17:17
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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I am still with 1.2.13, but can try 1.2.14 later if it helps. What you can always do is sniffing the net to see what events Asterisk sends to determine whether its a problem that can be fixed in Asterisk-Java (event syntax changed and event isnt recognized) or Asterisk (event isnt sent at all). =Stefan Mattias Malmquist wrote: > Hi, > > Since upgrading from Asterisk 1.2.7 to 1.2.14 I suddenly don't recieve > the ExtensionStatusEvent any longer. I have tried version 1.2.13 and did > receive them there. Do any of you know of any changes between these > versions that may trigger this behaviour? Do any of you have a solution > for the problem? > > I know this is not an Asterisk-Java specific question, but I figured it > might be of interest to someone else at this forum... > > Best regards > > Mattias Malmquist - Engineering, wx3 Telecom AB > > > > ------------------------------------------------------------------------- > Using Tomcat but need to do more? Need to support web services, security? > Get stuff done quickly with pre-integrated technology to make your job easier. > Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo > http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFybUlcVCZDfrn+pMRAhP6AJ4/cNkoTy8LDVRBFQXF/XoHph8CyACcCByo V9VlCuUw5iEWFJJGmXplJd4= =49fB -----END PGP SIGNATURE----- |
From: Mattias M. <ma...@wx...> - 2007-02-06 09:23:29
|
Hi, Since upgrading from Asterisk 1.2.7 to 1.2.14 I suddenly don't recieve the ExtensionStatusEvent any longer. I have tried version 1.2.13 and did receive them there. Do any of you know of any changes between these versions that may trigger this behaviour? Do any of you have a solution for the problem? I know this is not an Asterisk-Java specific question, but I figured it might be of interest to someone else at this forum... Best regards Mattias Malmquist - Engineering, wx3 Telecom AB=20 |
From: Stefan R. <ste...@re...> - 2007-02-03 16:53:59
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> I see that AsteriskServerImpl upon reconnects to the Asterisk server wi= ll > clears all the Channels in memory and do nothing about the > originateCallBacks (it might have quite a number of callbacks registere= d > and they won't be freed until removed from originateCallbacks). >=20 > l think we should only remove channels that are no longer returned by > StatusAction() and also remove the callbacks that don't have any > corresponding channels. This one is really tricky. You are right that a bunch of bugs are triggered in the live package when a reconnect occurs. This mostly results in memory leaks due to listeners not being removed. Besides that the current reconnect behaviour is far too simplistic. It simply removes all live objects and creates new ones based on Asterisk's current state. What can really happen to a LiveObject while we were disconnected is: a) it has been removed (user hang up, config was changed so that a queue no longer exists and so on) b) state has changed (channel was linked to another, ...) c) a new one has been created (new channel, etc.) I think we have to somehow inform the users of the library of these changes so they can act appropriatly. This is currently not done at all resulting in lots of logic needed within the client to support reconnect situations. Especially a) and c) are tricky here: We don't receive all state changes (e.g. extensions a channel has visited while we were disconnected) and the time a channel was removed (which might be interesting to clients that do accounting based on the state changes they receive). c) is less problematic as it equals the situation we have when we initially connect to Asterisk. What we have though is a new and consistent state that we retrieve from Asterisk (e.g. the result of the StatusAction for channels). So we could synthesize events based on a diff of the state before we got disconnected and the current state. These synthesized events could be delivered to the client through the listener approach we already have in place. As far as I can see this is not a trivial task at all, but I don't see much alternatives. I would be very glad to receive your thoughts on this 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... |
From: Stefan R. <ste...@re...> - 2007-02-03 16:37:22
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King Ho wrote: > The attached patch, against trunk, will add a shutdown() method to > AsteriskServer which will in turn shutdown the threads in > ManagerEventListenerProxy if necessary. You can use it however you like= or > to throw it away if you don't like it;) thanks for the patch! I've applied it as provided. =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... |
From: Stefan R. <ste...@re...> - 2007-02-03 14:47:43
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Hi, > ManagerConnection.sendEventGeneratingAction() and OriginateAction. > After receiving the corresponding CdrEvent we disconnect it with logoff= > method. The problem is that this logoff throws an Exception: >=20 > net.sf.asterisk.manager.TimeoutException: Timeout waiting for response = to > Logoff at > net.sf.asterisk.manager.DefaultManagerConnection.sendAction(DefaultMana= gerCo > nnection.java:634) at > net.sf.asterisk.manager.DefaultManagerConnection.sendAction(DefaultMana= gerCo > nnection.java:600) at > net.sf.asterisk.manager.DefaultManagerConnection.logoff >=20 > As we could verify, the Manager Connection was logged off in Asterisk, = but > in the Framework the ManagerConnection are still alive. >=20 > Any idea or suggestion? Can you provide a network dump of what's going on there? Use ngrep to sniff the network traffic to/from Asterisk's Manager API. You can either run ngrep on the Asterisk box or on the client that runs the Asterisk-Java based application. ngrep -s 4000 port 5038 -T =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... |
From: Gabriel A. <gab...@gl...> - 2007-02-02 19:52:09
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Hi All, We're making outgoing calls using ManagerConnection.sendEventGeneratingAction() and OriginateAction. After receiving the corresponding CdrEvent we disconnect it with logoff method. The problem is that this logoff throws an Exception: net.sf.asterisk.manager.TimeoutException: Timeout waiting for response to Logoff at net.sf.asterisk.manager.DefaultManagerConnection.sendAction(DefaultManagerCo nnection.java:634) at net.sf.asterisk.manager.DefaultManagerConnection.sendAction(DefaultManagerCo nnection.java:600) at net.sf.asterisk.manager.DefaultManagerConnection.logoff As we could verify, the Manager Connection was logged off in Asterisk, but in the Framework the ManagerConnection are still alive. Any idea or suggestion? Thanks in advance, Gabriel. |
From: David R. <dr...@in...> - 2007-02-01 17:05:47
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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 =C5=81ukasz Zaj=C4=85c wrote: > I need to match those events, because I want to know why the call faile= d. > I've spend a couple of days trying to solve that issue, so I'd really > appreciate some advice. I've had a similar problem (I needed to match OriginateActions with the NewChannel/CallerID/StateEvents it generates). I solved it by setting a variable on the OriginateAction. The generated channel (which generates the HangupEvent) should then have this variable set. Hope that helps, David -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFwh14ah2J2g3uuwIRArS1AKCL0sSGONptcblpKCXi+jQz6WeLswCfcw0I b5GKb64ScEjC7vCtkPyM1c0=3D =3DKiam -----END PGP SIGNATURE----- |