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From: Stefan R. <ste...@re...> - 2006-10-26 08:25:58
|
g f wrote: > What is the syntax for the above method so that the call will be comple= ted? > I tried the actual number but that doesnt work, I also tried > iax2/918504997800 but that doesnt work. I'd suggest you use the context/extension/priority properties of OriginateAction for your destination. You must not mix context/extension/priority and application/data - they are mutually exclusive. =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...> - 2006-10-26 08:23:35
|
Paulo Vieira Jr wrote: > 1 - Is there some way to play a file in a meetme? You can use the Manager API to originate a call from a Local channel that points somewhere in the dialplan to a Play() command and join that channel into meetme. =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...> - 2006-10-26 08:22:40
|
Bruno Negrao wrote: > But I'd like to confirm with you if you agree that patching asterisk is= > the best solution for this problem. Yes the patch is a good thing. Remember that you must first send a disclaimer to Digium for the patch to be included in the main Asterisk distribution. Its described on mantis= =2E =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: Bruno N. <bn...@gm...> - 2006-10-24 19:09:56
|
Stefan, I can prepare a patch for asterisk and fill a bug report. And I believe this is the best solution. But I'd like to confirm with you if you agree that patching asterisk is the best solution for this problem. I'm asking your confirmation because your opinion will endorse me when I'll talk about this step with my customer. Thank you, Bruno. On 10/23/06, Stefan Reuter <ste...@re...> wrote: > > Bruno Negrao wrote: > > I started discussing this topic at the Asterisk's Developers forum: > > > > http://forums.digium.com/viewtopic.php?t=10578 > > if you already have a patch it might be better to just open a bug on > mantis and post the patch so it gets included in the next version of > Asterisk. > > =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... > > > > ------------------------------------------------------------------------- > 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 > > > > |
|
From: <fr...@pb...> - 2006-10-24 18:41:23
|
> type show manager actions on Asterisk's CLI to see if it supports that > action Thanks, found it in version 1.4.0-beta3, so voip-info was wrong. Poul |
|
From: Stefan R. <ste...@re...> - 2006-10-23 21:18:41
|
Bruno Negrao wrote: > I started discussing this topic at the Asterisk's Developers forum: >=20 > http://forums.digium.com/viewtopic.php?t=3D10578 if you already have a patch it might be better to just open a bug on mantis and post the patch so it gets included in the next version of Asterisk. =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...> - 2006-10-23 21:17:37
|
Poul M=F8ller Hansen wrote: > But I get this returned "Error Invalid/unknown command" >=20 > According to the Asterisk dokumentation the command should be available= =20 > as of ver. 1.2.8 ?? type show manager actions on Asterisk's CLI to see if it supports that action. =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: g f <gf...@gm...> - 2006-10-23 20:56:17
|
Hello all,
currently I am using asterisk-java 0-2.jar with trixbox 1.2
I am successful at making sip-sip calls using the following code:
public void callUser(String callee, String caller, String callerID) throws
IOException,
AuthenticationFailedException, TimeoutException
{
OriginateAction oAction;
ManagerResponse mResponse;
oAction = new OriginateAction();
oAction.setApplication("dial");
oAction.setChannel(caller); //this is the person that is calling FORMAT
sip/201.
oAction.setData(callee); //this is the person that is being called.
FORMAT sip/201
oAction.setCallerId(callerID); //this is the callerID of the person
calling.
oAction.setPriority(new Integer(1));
oAction.setTimeout(new Long(30000));
oAction.setAsync(Boolean.TRUE);
As you can see I pass the following strings for callee and caller : sip/201,
etc...
Now I have an IAX2 trunk using voipjet.
I can call 918504997800 from a xlite softphone and trixbox recognizes this
as an outbound line (bc of the 9) and sends it to the iax2 trunk and all
works well.
What is the syntax for the above method so that the call will be completed?
I tried the actual number but that doesnt work, I also tried
iax2/918504997800 but that doesnt work.
Any ideas?
Thanks!
|
|
From: <fr...@pb...> - 2006-10-23 20:39:15
|
Hi,
has anyone of you succeded in using PlayDtmfAction(); ?
I have tried this against an Asterisk ver. 1.2.13 server:
playdtmfaction = new PlayDtmfAction();
playdtmfaction.setChannel("SIP/pmh");
playdtmfaction.setDigit(number);
playdtmfresponse = connection.sendAction(playdtmfaction, 30000);
System.out.println(playdtmfresponse.getResponse()+"
"+playdtmfresponse.getMessage());
But I get this returned "Error Invalid/unknown command"
According to the Asterisk dokumentation the command should be available
as of ver. 1.2.8 ??
Thanks,
Poul
|
|
From: Paulo V. J. <pau...@po...> - 2006-10-23 15:21:53
|
Hi all,
I have two questions:
1 - Is there some way to play a file in a meetme?
2 - I'm starting a meetme using command:
channel.sendCommand("meetme", "1234|p");
Is there another way to create the conference and add users?
I'm using 0.3m1.
Thanks.
Paulo.
|
|
From: Bruno N. <bn...@gm...> - 2006-10-23 14:15:46
|
Stefan, I started discussing this topic at the Asterisk's Developers forum: http://forums.digium.com/viewtopic.php?t=10578 Regards, Bruno |
|
From: Stefan R. <ste...@re...> - 2006-10-23 10:50:02
|
Jonathan Augenstine wrote: > I understand that OpenPBX is a branch of Asterisk. Will Asterisk-=20 > Java function with OpenPBX? I have no idea. If you try it please report your results. =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: Jonathan A. <jau...@st...> - 2006-10-23 03:36:16
|
I understand that OpenPBX is a branch of Asterisk. Will Asterisk- Java function with OpenPBX? Jonathan |
|
From: Yelson V. <yv...@gm...> - 2006-10-20 23:50:44
|
Hi Guys i need to create a program that can be able to send a messsage to a fastagi, but i've been trying using regular socket (DataInputStream(sk.getInputStream()) .... etc) but i don't known how can i send the fastagi parameters (the fastagi receive the socket fine i can see it in the debug console but i don't know how to send the parameters) ??? i tried using url, but says that "agi" is not a protocol supported, So any idea will be welcome Thanks -- Yelson E Vivas C MPC (571) 6500-800 |
|
From: King H. <kin...@ne...> - 2006-10-20 10:12:31
|
Sorry, I missed this:
public ManagerEventListenerProxy()
{
this.executor =3D Executors.newSingleThreadExecutor(new
DaemonThreadFactory());
}
So the handling of event is serialized. Please ignore my previous post!!
Best Regards,
King
-----=AD=EC=A9l=B6l=A5=F3-----
=B1H=A5=F3=AA=CC: ast...@li...
[mailto:ast...@li...] =A5N=B2z King =
Ho
=B1H=A5=F3=A4=E9=B4=C1: Friday, 20 October, 2006 12:03
=A6=AC=A5=F3=AA=CC: ast...@li...
=A5D=A6=AE: Re: [Asterisk-java-users] Strange problem with =
originateToExtension
Hi,
After looking at the code more in detail, it seems that
DefaultAsteriskServer will use ManagerEventListenerProxy to handle the
dispatch of events asynchronously. However, it looks like the
onManagerEvent() of ManagerEventListenerProxy will create a separate =
thread
for each event so that the system may finish handling a later arrive =
event
earlier than a previously arrive event.
I think having each event being handled in a separate thread may create
problem in saturations where a later arrived event depends on the =
complete
processing of a previously arrived event.
I think one way to achive serialization of the event handling is to =
append
the events in a queue in onManagerEvent() of ManagerEventListenerProxy. =
And
ManagerEventListenerProxy (probably make it a Runnable) will run a =
separate
thread getting events from the queue and call =
target.onManagerEvent(event).
This is one way I can think of that will do it but may not be the best =
way.
Thanks.
Best Regards,
King
-----=AD=EC=A9l=B6l=A5=F3-----
=B1H=A5=F3=AA=CC: ast...@li...
[mailto:ast...@li...] =A5N=B2z King =
Ho
=B1H=A5=F3=A4=E9=B4=C1: Thursday, 19 October, 2006 19:03
=A6=AC=A5=F3=AA=CC: ast...@li...
=A5D=A6=AE: [Asterisk-java-users] Strange problem with =
originateToExtension
Hi,
I am having a very strange problem (probably a timing problem) with
asterisk-java-m1. I am using the =
DefaultAsteriskServer.originateToExtension
method as follows:
defaultAsteriskServer =3D new
DefaultAsteriskServer("192.168.0.238", "testing", "testing123");
defaultAsteriskServer.initialize();
=20
defaultAsteriskServer.originateToExtension("SIP/1000@ray_server","testing=
","
1003",1,30000);
The call did go through OK but there is a NoSuchChannelException thown.
Below is the console output along with stack trace from the exception:
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Connecting to 192.168.0.238:5038
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Connected via Asterisk Call Manager/1.0
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Successfully logged in
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Determined Asterisk version: Asterisk 1.2
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Initializing done
org.asteriskjava.live.NoSuchChannelException: Channel =
'SIP/1000@ray_server'
is not available
at
org.asteriskjava.live.internal.AsteriskServerImpl.originate(AsteriskServe=
rIm
pl.java:339)
at
org.asteriskjava.live.internal.AsteriskServerImpl.originateToExtension(As=
ter
iskServerImpl.java:278)
at
org.asteriskjava.live.internal.AsteriskServerImpl.originateToExtension(As=
ter
iskServerImpl.java:258)
at
org.asteriskjava.live.DefaultAsteriskServer.originateToExtension(DefaultA=
ste
riskServer.java:133)
at hk.com.csl.asterisk_java_example.Manager.run(Manager.java:29)
at hk.com.csl.asterisk_java_example.Manager.main(Manager.java:40)
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Adding channel SIP/ray_server-09e11ee0(1161253134.58)
Based on the console output, the problem seems to be that the =
"Newchannel"
event was processed after the "OriginateSuccess" event so when it tried =
to
lookup the channel, it wasn't added yet.
I have connected to the Asterisk Manager port and look at the order in =
which
the events are happening and Newchannel is happening before =
OriginateSuccess
which is correct.
The version of Asterisk that I am using is 1.2.12.1 and also tried with =
1.2.
13 with the same problem.
Thanks.
Best Regards,
King
__________ NOD32 1.1810 (20061018) Information __________
This message was checked by NOD32 antivirus system.
http://www.nod32.com.hk
__________ NOD32 1.1810 (20061018) Information __________
This message was checked by NOD32 antivirus system.
http://www.nod32.com.hk
|
|
From: King H. <kin...@ne...> - 2006-10-20 06:47:16
|
Sorry, I missed this:
public ManagerEventListenerProxy()
{
this.executor =3D Executors.newSingleThreadExecutor(new
DaemonThreadFactory());
}
So the handling of event is serialized. Please ignore my previous post!!
Best Regards,
King
-----=AD=EC=A9l=B6l=A5=F3-----
=B1H=A5=F3=AA=CC: ast...@li...
[mailto:ast...@li...] =A5N=B2z King =
Ho
=B1H=A5=F3=A4=E9=B4=C1: Friday, 20 October, 2006 12:03
=A6=AC=A5=F3=AA=CC: ast...@li...
=A5D=A6=AE: Re: [Asterisk-java-users] Strange problem with =
originateToExtension
Hi,
After looking at the code more in detail, it seems that
DefaultAsteriskServer will use ManagerEventListenerProxy to handle the
dispatch of events asynchronously. However, it looks like the
onManagerEvent() of ManagerEventListenerProxy will create a separate =
thread
for each event so that the system may finish handling a later arrive =
event
earlier than a previously arrive event.
I think having each event being handled in a separate thread may create
problem in saturations where a later arrived event depends on the =
complete
processing of a previously arrived event.
I think one way to achive serialization of the event handling is to =
append
the events in a queue in onManagerEvent() of ManagerEventListenerProxy. =
And
ManagerEventListenerProxy (probably make it a Runnable) will run a =
separate
thread getting events from the queue and call =
target.onManagerEvent(event).
This is one way I can think of that will do it but may not be the best =
way.
Thanks.
Best Regards,
King
-----=AD=EC=A9l=B6l=A5=F3-----
=B1H=A5=F3=AA=CC: ast...@li...
[mailto:ast...@li...] =A5N=B2z King =
Ho
=B1H=A5=F3=A4=E9=B4=C1: Thursday, 19 October, 2006 19:03
=A6=AC=A5=F3=AA=CC: ast...@li...
=A5D=A6=AE: [Asterisk-java-users] Strange problem with =
originateToExtension
Hi,
I am having a very strange problem (probably a timing problem) with
asterisk-java-m1. I am using the =
DefaultAsteriskServer.originateToExtension
method as follows:
defaultAsteriskServer =3D new
DefaultAsteriskServer("192.168.0.238", "testing", "testing123");
defaultAsteriskServer.initialize();
=20
defaultAsteriskServer.originateToExtension("SIP/1000@ray_server","testing=
","
1003",1,30000);
The call did go through OK but there is a NoSuchChannelException thown.
Below is the console output along with stack trace from the exception:
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Connecting to 192.168.0.238:5038
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Connected via Asterisk Call Manager/1.0
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Successfully logged in
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Determined Asterisk version: Asterisk 1.2
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Initializing done
org.asteriskjava.live.NoSuchChannelException: Channel =
'SIP/1000@ray_server'
is not available
at
org.asteriskjava.live.internal.AsteriskServerImpl.originate(AsteriskServe=
rIm
pl.java:339)
at
org.asteriskjava.live.internal.AsteriskServerImpl.originateToExtension(As=
ter
iskServerImpl.java:278)
at
org.asteriskjava.live.internal.AsteriskServerImpl.originateToExtension(As=
ter
iskServerImpl.java:258)
at
org.asteriskjava.live.DefaultAsteriskServer.originateToExtension(DefaultA=
ste
riskServer.java:133)
at hk.com.csl.asterisk_java_example.Manager.run(Manager.java:29)
at hk.com.csl.asterisk_java_example.Manager.main(Manager.java:40)
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Adding channel SIP/ray_server-09e11ee0(1161253134.58)
Based on the console output, the problem seems to be that the =
"Newchannel"
event was processed after the "OriginateSuccess" event so when it tried =
to
lookup the channel, it wasn't added yet.
I have connected to the Asterisk Manager port and look at the order in =
which
the events are happening and Newchannel is happening before =
OriginateSuccess
which is correct.
The version of Asterisk that I am using is 1.2.12.1 and also tried with =
1.2.
13 with the same problem.
Thanks.
Best Regards,
King
__________ NOD32 1.1810 (20061018) Information __________
This message was checked by NOD32 antivirus system.
http://www.nod32.com.hk
__________ NOD32 1.1810 (20061018) Information __________
This message was checked by NOD32 antivirus system.
http://www.nod32.com.hk
|
|
From: King H. <kin...@ne...> - 2006-10-20 04:03:48
|
Hi,
After looking at the code more in detail, it seems that
DefaultAsteriskServer will use ManagerEventListenerProxy to handle the
dispatch of events asynchronously. However, it looks like the
onManagerEvent() of ManagerEventListenerProxy will create a separate =
thread
for each event so that the system may finish handling a later arrive =
event
earlier than a previously arrive event.
I think having each event being handled in a separate thread may create
problem in saturations where a later arrived event depends on the =
complete
processing of a previously arrived event.
I think one way to achive serialization of the event handling is to =
append
the events in a queue in onManagerEvent() of ManagerEventListenerProxy. =
And
ManagerEventListenerProxy (probably make it a Runnable) will run a =
separate
thread getting events from the queue and call =
target.onManagerEvent(event).
This is one way I can think of that will do it but may not be the best =
way.
Thanks.
Best Regards,
King
-----=AD=EC=A9l=B6l=A5=F3-----
=B1H=A5=F3=AA=CC: ast...@li...
[mailto:ast...@li...] =A5N=B2z King =
Ho
=B1H=A5=F3=A4=E9=B4=C1: Thursday, 19 October, 2006 19:03
=A6=AC=A5=F3=AA=CC: ast...@li...
=A5D=A6=AE: [Asterisk-java-users] Strange problem with =
originateToExtension
Hi,
I am having a very strange problem (probably a timing problem) with
asterisk-java-m1. I am using the =
DefaultAsteriskServer.originateToExtension
method as follows:
defaultAsteriskServer =3D new
DefaultAsteriskServer("192.168.0.238", "testing", "testing123");
defaultAsteriskServer.initialize();
=20
defaultAsteriskServer.originateToExtension("SIP/1000@ray_server","testing=
","
1003",1,30000);
The call did go through OK but there is a NoSuchChannelException thown.
Below is the console output along with stack trace from the exception:
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Connecting to 192.168.0.238:5038
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Connected via Asterisk Call Manager/1.0
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Successfully logged in
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Determined Asterisk version: Asterisk 1.2
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Initializing done
org.asteriskjava.live.NoSuchChannelException: Channel =
'SIP/1000@ray_server'
is not available
at
org.asteriskjava.live.internal.AsteriskServerImpl.originate(AsteriskServe=
rIm
pl.java:339)
at
org.asteriskjava.live.internal.AsteriskServerImpl.originateToExtension(As=
ter
iskServerImpl.java:278)
at
org.asteriskjava.live.internal.AsteriskServerImpl.originateToExtension(As=
ter
iskServerImpl.java:258)
at
org.asteriskjava.live.DefaultAsteriskServer.originateToExtension(DefaultA=
ste
riskServer.java:133)
at hk.com.csl.asterisk_java_example.Manager.run(Manager.java:29)
at hk.com.csl.asterisk_java_example.Manager.main(Manager.java:40)
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Adding channel SIP/ray_server-09e11ee0(1161253134.58)
Based on the console output, the problem seems to be that the =
"Newchannel"
event was processed after the "OriginateSuccess" event so when it tried =
to
lookup the channel, it wasn't added yet.
I have connected to the Asterisk Manager port and look at the order in =
which
the events are happening and Newchannel is happening before =
OriginateSuccess
which is correct.
The version of Asterisk that I am using is 1.2.12.1 and also tried with =
1.2.
13 with the same problem.
Thanks.
Best Regards,
King
__________ NOD32 1.1810 (20061018) Information __________
This message was checked by NOD32 antivirus system.
http://www.nod32.com.hk
|
|
From: Bruno N. <bn...@gm...> - 2006-10-19 23:58:10
|
> Besides that it would be interesting to find out what is causing the > problem. And either fix Asterisk or Asterisk-Java. If you dont mind > please connect to the Manager API through telnet or sniff the traffic > via ngrep and compare the events to the state changes in Asterisk-Java. > If you see everything is in parallel we have a problem in Asterisk, if > not indicate which events were not correctly handled and i will fix it. Stefan, I did the test using a sniffer and I concluded that asterisk is not reporting the latest events that would set the AsteriskChannels in a consistent state. Specifically, asterisk is omitting a Newcallerid event for the renamed channel that remained active, and is omitting two Newaccount (this event does not exist yet) for the two channels that remained active. The latest events sent by asterisk paralleled with the PropertyChangeEvents are described bellow. A file with the full paralleled events is attached. So what can we do now? How to report this for the asterisk staff? Would they ever listen to us? thank you, bruno. Event: Rename Privilege: call,all Oldname: SIP/3026-0a1e Newname: SIP/3026-0a1e<MASQ> Uniqueid: 1161291777.920 Server: localhost Property change (name): AsteriskChannel[id='1161291777.920', name='SIP/3026-0a1e<MASQ>',callerId='<3026>',state='UP', account='null',dateOfCreation=Thu Oct 19 18:04:40 BRST 2006, dialedChannel=null,dialingChannel=AsteriskChannel[id='1161291771.917', name='Local/3025@agente_out-6caa,1'], linkedChannel=AsteriskChannel[id='1161291771.917', name='Local/3025@agente_out-6caa,1']] 4877503 Event: Rename Privilege: call,all Oldname: Local/3025@agente_out-6caa,2 Newname: SIP/3026-0a1e Uniqueid: 1161291771.918 Server: localhost Property change (name): AsteriskChannel[id='1161291771.918',name='SIP/3026-0a1e',callerId='<3025>', state='UP',account='null',dateOfCreation=Thu Oct 19 18:04:34 BRST 2006, dialedChannel=AsteriskChannel[id='1161291771.919',name='SIP/3025-0c41'], dialingChannel=null,linkedChannel=AsteriskChannel[id='1161291771.919', name='SIP/3025-0c41']] 10605044 Event: Rename Privilege: call,all Oldname: SIP/3026-0a1e<MASQ> Newname: Local/3025@agente_out-6caa,2<ZOMBIE> Uniqueid: 1161291777.920 Server: localhost Property change (name): AsteriskChannel[id='1161291777.920',name='Local/3025@agente_out-6caa,2<ZOMBIE>', callerId='<3026>',state='UP',account='null',dateOfCreation=Thu Oct 19 18:04:40 BRST 2006, dialedChannel=null, dialingChannel=AsteriskChannel[id='1161291771.917', name='Local/3025@agente_out-6caa,1'],linkedChannel=AsteriskChannel[id='1161291771.917',name='Local/3025@agente_out-6caa,1']] 4877503 Event: Unlink Privilege: call,all Channel1: Local/3025@agente_out-6caa,1 Channel2: Local/3025@agente_out-6caa,2<ZOMBIE> Uniqueid1: 1161291771.917 Uniqueid2: 1161291777.920 CallerID1: 3025 CallerID2: 3025 Server: localhost Property change (linkedChannel): AsteriskChannel[id='1161291771.917',name='Local/3025@agente_out-6caa,1',callerId='<3025>',state='UP',account='null',dateOfCreation=Thu Oct 19 18:04:34 BRST 2006, dialedChannel=AsteriskChannel[id='1161291777.920', name='Local/3025@agente_out-6caa,2<ZOMBIE>'],dialingChannel=null,linkedChannel=null] 14440411 Property change (linkedChannel): AsteriskChannel[id='1161291777.920',name='Local/3025@agente_out-6caa, 2<ZOMBIE>', callerId='<3026>',state='UP',account='null', dateOfCreation=Thu Oct 19 18:04:40 BRST 2006, dialedChannel=null,dialingChannel=AsteriskChannel[id='1161291771.917', name='Local/3025@agente_out-6caa,1'],linkedChannel=null] 4877503 Event: Hangup Privilege: call,all Channel: Local/3025@agente_out-6caa,2<ZOMBIE> Uniqueid: 1161291777.920 Cause: 16 Cause-txt: Normal Clearing Server: localhost Property change (state): AsteriskChannel[id='1161291777.920',name='Local/3025@agente_out-6caa,2<ZOMBIE>',callerId='<3026>',state='HUNGUP',account='null',dateOfCreation=Thu Oct 19 18:04:40 BRST 2006,dialedChannel=null,dialingChannel=AsteriskChannel[id='1161291771.917',name='Local/3025@agente_out-6caa,1'],linkedChannel=null] 4877503 Event: Hangup Privilege: call,all Channel: Local/3025@agente_out-6caa,1 Uniqueid: 1161291771.917 Cause: 16 Cause-txt: Normal Clearing Server: localhost Property change (state): AsteriskChannel[id='1161291771.917',name='Local/3025@agente_out-6caa,1',callerId='<3025>',state='HUNGUP',account='null',dateOfCreation=Thu Oct 19 18:04:34 BRST 2006,dialedChannel=AsteriskChannel[id='1161291777.920',name='Local/3025@agente_out-6caa,2<ZOMBIE>'],dialingChannel=null,linkedChannel=null] 14440411 |
|
From: King H. <kin...@ne...> - 2006-10-19 11:03:25
|
Hi,
I am having a very strange problem (probably a timing problem) with
asterisk-java-m1. I am using the =
DefaultAsteriskServer.originateToExtension
method as follows:
defaultAsteriskServer =3D new
DefaultAsteriskServer("192.168.0.238", "testing", "testing123");
defaultAsteriskServer.initialize();
=20
defaultAsteriskServer.originateToExtension("SIP/1000@ray_server","testing=
","
1003",1,30000);
The call did go through OK but there is a NoSuchChannelException thown.
Below is the console output along with stack trace from the exception:
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Connecting to 192.168.0.238:5038
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Connected via Asterisk Call Manager/1.0
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Successfully logged in
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Determined Asterisk version: Asterisk 1.2
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Initializing done
org.asteriskjava.live.NoSuchChannelException: Channel =
'SIP/1000@ray_server'
is not available
at
org.asteriskjava.live.internal.AsteriskServerImpl.originate(AsteriskServe=
rIm
pl.java:339)
at
org.asteriskjava.live.internal.AsteriskServerImpl.originateToExtension(As=
ter
iskServerImpl.java:278)
at
org.asteriskjava.live.internal.AsteriskServerImpl.originateToExtension(As=
ter
iskServerImpl.java:258)
at
org.asteriskjava.live.DefaultAsteriskServer.originateToExtension(DefaultA=
ste
riskServer.java:133)
at hk.com.csl.asterisk_java_example.Manager.run(Manager.java:29)
at hk.com.csl.asterisk_java_example.Manager.main(Manager.java:40)
2006=A6~10=A4=EB19=A4=E9 =A4U=A4=C806:19:40 =
org.asteriskjava.util.internal.JavaLoggingLog
info
=B8=EA=B0T: Adding channel SIP/ray_server-09e11ee0(1161253134.58)
Based on the console output, the problem seems to be that the =
"Newchannel"
event was processed after the "OriginateSuccess" event so when it tried =
to
lookup the channel, it wasn't added yet.
I have connected to the Asterisk Manager port and look at the order in =
which
the events are happening and Newchannel is happening before =
OriginateSuccess
which is correct.
The version of Asterisk that I am using is 1.2.12.1 and also tried with =
1.2.
13 with the same problem.
Thanks.
Best Regards,
King
|
|
From: Johannes B. <j....@ad...> - 2006-10-18 13:52:01
|
Okay ... it was my fault. My calls store some details with the = channel-names in it and I accidentally used channel.toString() in one = place instead of channel.getName(). I still get hangups on redirect but at least it is done without an error = now. |
|
From: Johannes B. <j....@ad...> - 2006-10-17 01:27:42
|
Well, I think the issue has to be located somewhere else. I've patched = my bristuff and tried it on a fresh trixbox install, too. The same = result: it keeps telling me one of the channels does not exist. In case = I try redirectAction with two channels I'm told the second channel = (connected to Zap/1 e.g.) does not exist. And I'm always disconnected = afterwards. Well I can't remember more details at the moment. As to asterisk-jtapi ... it's not related to that. I gave it a try once = and couldn't get it running so I sticked to the asterisk-provider I = checked into GJTapi-cvs a year or so ago. That one worked fine for me so = far and that's why I'm continuing with it. I don't want to offend = someone but besides the GJTapi project is a better place for an = GJTapi-provider than a separate sourceforge project I think. Johannes Boesl |
|
From: <da...@re...> - 2006-10-16 23:22:20
|
VGhhbmtzLCBTdGVmYW4uICBJIGhhZCBzdXNwZWN0ZWQgYXMgbXVjaC4gIEknbGwgaW5jbHVkZSBh biBBcGFjaGUgaW5zdGFsbCB3LyB0aGUgc29mdHdhcmUuDQoNCg0KU2VudCB2aWEgQmxhY2tCZXJy eSBmcm9tIENpbmd1bGFyIFdpcmVsZXNzICANCg0KLS0tLS1PcmlnaW5hbCBNZXNzYWdlLS0tLS0N CkZyb206IFN0ZWZhbiBSZXV0ZXIgPHN0ZWZhbi5yZXV0ZXJAcmV1Y29uLmNvbT4NCkRhdGU6IFR1 ZSwgMTcgT2N0IDIwMDYgMDE6MTQ6MDIgDQpUbzphc3Rlcmlzay1qYXZhLXVzZXJzQGxpc3RzLnNv dXJjZWZvcmdlLm5ldA0KU3ViamVjdDogUmU6IFtBc3Rlcmlzay1qYXZhLXVzZXJzXSBTdHJlYW1p bmcgYW5kIHJlY29yZGluZyByZW1vdGUgZmlsZXMNCiB0aHJvdWdoIEFHST8NCg0KSGkgRGF2ZSwN Cg0KPiBTbywgSSdkIGxpa2UgdG8gYmUgYWJsZSB0byB0cmFuc2ZlciB0aGUgcmVjb3JkZWQgZmls ZXMgZnJvbSB0aGUgQXN0ZXJpc2sNCj4gc2VydmVyIHRvIG15IEphdmEgc2VydmVyLiAgSXMgdGhl cmUgYW55IGJ1aWx0LWluIHdheSB0byBkbyB0aGlzLCBvciBkbyBJDQo+IG5lZWQgdG8gaW5jbHVk ZSBhIHdlYiBzZXJ2ZXIgb3IgRlRQIHNlcnZlciBvbiB0aGUgQXN0ZXJpc2sgc2VydmVyPw0KDQpV bmZvcnR1bmF0ZWx5IHRoaXMgaXMgbm90aGluZyB0aGUgTWFuYWdlciBBUEkgb3IgRmFzdEFHSSBw cm92aWRlIHNvIHlvdQ0KaGF2ZSB0byBzdGljayB0byBzb21lIG90aGVyIG1lYW5zLg0KV2hhdCBw ZW9wbGUgdXN1YWxseSBkbyBpcyBlaXRoZXIgcGxhY2luZyB0aGUgZmlsZXMgb24gYW4gTkZTIHNo YXJlIG9yDQp1c2luZyBzb21lIEphdmEgYmFzZWQgU1NIIGNsaWVudCBsaWIgZm9yIGZpbGUgdHJh bnNmZXIsIGUuZy4NCmh0dHA6Ly93d3cuZ2FueW1lZC5ldGh6LmNoL3NzaDIvDQoNCj1TdGVmYW4N Cg0KLS0gDQpyZXV0ZXIgbmV0d29yayBjb25zdWx0aW5nDQpOZXVzc2VyIFN0ci4gMTEwDQo1MDc2 MCBLb2Vsbg0KR2VybWFueQ0KVGVsZWZvbjogKzQ5IDIyMSAxMzA1Njk5LTANClRlbGVmYXg6ICs0 OSAyMjEgMTMwNTY5OS05MA0KRS1NYWlsOiAgc3RlZmFuLnJldXRlckByZXVjb24uY29tDQpKYWJi ZXI6ICBzdGVmYW4ucmV1dGVyQHJldWNvbi5jb20NCg0KDQotLS0tLS0tLS0tLS0tLS0tLS0tLS0t LS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tDQpVc2lu ZyBUb21jYXQgYnV0IG5lZWQgdG8gZG8gbW9yZT8gTmVlZCB0byBzdXBwb3J0IHdlYiBzZXJ2aWNl cywgc2VjdXJpdHk/DQpHZXQgc3R1ZmYgZG9uZSBxdWlja2x5IHdpdGggcHJlLWludGVncmF0ZWQg dGVjaG5vbG9neSB0byBtYWtlIHlvdXIgam9iIGVhc2llcg0KRG93bmxvYWQgSUJNIFdlYlNwaGVy ZSBBcHBsaWNhdGlvbiBTZXJ2ZXIgdi4xLjAuMSBiYXNlZCBvbiBBcGFjaGUgR2Vyb25pbW8NCmh0 dHA6Ly9zZWwuYXMtdXMuZmFsa2FnLm5ldC9zZWw/Y21kPWxuayZraWQ9MTIwNzA5JmJpZD0yNjMw NTcmZGF0PTEyMTY0Mg0KX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19f X19fX18NCkFzdGVyaXNrLWphdmEtdXNlcnMgbWFpbGluZyBsaXN0DQpBc3Rlcmlzay1qYXZhLXVz ZXJzQGxpc3RzLnNvdXJjZWZvcmdlLm5ldA0KaHR0cHM6Ly9saXN0cy5zb3VyY2Vmb3JnZS5uZXQv bGlzdHMvbGlzdGluZm8vYXN0ZXJpc2stamF2YS11c2Vycw0KDQo= |
|
From: Stefan R. <ste...@re...> - 2006-10-16 23:18:18
|
Johannes Boesl wrote: > Indeed I'm using a bristuffed one. I've already read a posting from you= > somewhere regarding that bug (but somehow I thought it's not related to= > my problem O_o). I'll check that at once. Hehe I hope it solves the problem. If not feel free to tell us and I'll have a deeper look. The bristuff thing is just the first issue I thought of as it already caused a lot of headache for me, especially with new versions of Asterisk where I missed to apply it :) Just out of curioristy, is your JTAPI stuff related to Jens Wilke's project (http://asterisk-jtapi.sourceforge.net/) or different? =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...> - 2006-10-16 23:14:13
|
Hi Dave, > So, I'd like to be able to transfer the recorded files from the Asteris= k > server to my Java server. Is there any built-in way to do this, or do = I > need to include a web server or FTP server on the Asterisk server? Unfortunately this is nothing the Manager API or FastAGI provide so you have to stick to some other means. What people usually do is either placing the files on an NFS share or using some Java based SSH client lib for file transfer, e.g. http://www.ganymed.ethz.ch/ssh2/ =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...> - 2006-10-16 23:10:31
|
Did you enable the manager api in manager.conf? Did you make sure no firewall is preventing traffic to port 5039 (the manager api port) on the Asterisk box? Did you verify by telnet or some similar tool that you can actually access the manager api from the machine running your application? Note: The manager api uses a completly different protocol and connection than IAX2. =3DStefan Eric C. wrote: > I'm trying to use the HelloManager example from:=20 > http://asterisk-java.org/0.3-m1/tutorial.html but am not getting anywhe= re. >=20 > I have asterisk configured properly and it works whenever I put a *.cal= l=20 > file in the /var/spool/asterisk/outgoing directory. >=20 > My problem is that I get the error: > java.net.ConnectException: Connection refused >=20 > I have a softphone setup so that it can call the asterisk pbx and hear = the=20 > demo messages like the echo test and so on. >=20 > I have the following but do not know what is going on. >=20 > ManagerConnectionFactory factory =3D new=20 > ManagerConnectionFactory("192.168.0.100", "softPhoneUserName",=20 > "softPhonePassword"); >=20 >=20 > originateAction =3D new OriginateAction(); > originateAction.setChannel("IAX2/softPhoneUserName"); > originateAction.setContext("ContextProviderAsDefinedIn-IAX.CONF"); > originateAction.setExten("600"); > originateAction.setPriority(new Integer(1)); > originateAction.setTimeout(new Integer(30000)); >=20 >=20 > I really need help with this. >=20 > thanks, > Eric. >=20 >=20 >=20 > -----------------------------------------------------------------------= -- > Using Tomcat but need to do more? Need to support web services, securit= y? > 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 Geron= imo > http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat= =3D121642 > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users --=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... |
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