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From: Stefan R. <sr...@re...> - 2006-08-30 08:26:49
|
Chris Howard wrote: > I'm doing an originateToExtension using a local channel. I really need > to get the actual channel after the masquerading is complete. It seems= > that the masquerading takes a few milliseconds to complete so a delay > and was added until Local/10101@all_calls became SIP/joe_agent-2jd9.=20 > Can anyone think of a better way to do this. I really need to look at > the code that updates the channel as its fun to debug a channel while > the data values change while you are sitting at a breakpoint (as they > should when the status of the channel changes) You could use originateToExtensionAsync, provide a Callback and once the call is connected (i.e. inside the callback) register a PropertyChangeListener with the channel for the "name" property. This PropertyChangeListener is then called each time the name of this channel changes. Note that you should not do any heavy processing inside the callback and PropertyChangeListener as it's called from the same thread that updates other live objects and will effectivly block any other updates to be propagated (much like what you may be used to from swing). =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... |
|
From: Chris H. <ch...@as...> - 2006-08-30 05:04:18
|
I'm doing an originateToExtension using a local channel. I really
need to get the actual channel after the masquerading is complete.
It seems that the masquerading takes a few milliseconds to complete
so a delay and was added until Local/10101@all_calls became SIP/
joe_agent-2jd9. Can anyone think of a better way to do this. I
really need to look at the code that updates the channel as its fun
to debug a channel while the data values change while you are sitting
at a breakpoint (as they should when the status of the channel changes)
Thanks again!
------------------------------------------------------------------------
--------------------------
String device = "Local/"+agentExten+"@all_calls";
int loops = 0;
channel = myServer.originateToExtension(device, "all_calls",
remoteNumber, 1, new Long(30000));
while (channel.getName().startsWith(device) && loops < 1000)
{
loops++;
Thread.sleep(10);
}
------------------------------------------------------------------------
-------------------------- |
|
From: Chris H. <ch...@as...> - 2006-08-30 04:35:25
|
On Aug 29, 2006, at 9:31 PM, Thameem Ansari wrote: > I agree with Chris about versioning the release. As stefan said, > there is very small java community to develop applications for > asterisk but I feel that its more stable and enterprise standard if > you develop something in java. > The way I look at it, this 0.3 is really 2.0 :) Chris |
|
From: Thameem A. <tha...@ya...> - 2006-08-30 02:31:24
|
I agree with Chris about versioning the release. As stefan said, there is very small java community to develop applications for asterisk but I feel that its more stable and enterprise standard if you develop something in java. -Thameem Stefan Reuter <sr...@re...> wrote: Chris Howard wrote: > None (anymore). I migrated to 0.3 with very little problem. This > release seems to be solid. Thats good news. > On a side note, why are you sticking to the > 0.X numbering scheme. I think that this should really be 1.X by now. I think of the 1.0 release as being feature complete. For now the live package still misses functionalaity with regard to Queue and Member handling. Once this has been done I will consider a 1.0 release. An additional issue is maintainance: - As you have just noticed for now there is no real support for the "stable" release, i.e. bugs are fixed in the 0.3 milestone release but not in the "stable" 0.2. - A "stable" release should offer backwards compatibility. There were some very incompatible changes from 0.2 to 0.3 (with the change in the package name being the most visible one). In my opinion both would not be acceptable for a 1.x release. > It may get more use if it's not thought of as "Alpha" software. You might be right, though I notice that there is generally a very small Java community around Asterisk and we probably won't be able to convince any of the "script-kiddies" (with script being PHP/Perl/... ;) to switch to Java even with the greatest Asterisk support. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... ------------------------------------------------------------------------- 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 --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out. |
|
From: Stefan R. <sr...@re...> - 2006-08-29 21:03:09
|
Chris Howard wrote: > None (anymore). I migrated to 0.3 with very little problem. This=20 > release seems to be solid. Thats good news. > On a side note, why are you sticking to the=20 > 0.X numbering scheme. I think that this should really be 1.X by now. = I think of the 1.0 release as being feature complete. For now the live package still misses functionalaity with regard to Queue and Member handling. Once this has been done I will consider a 1.0 release. An additional issue is maintainance: - As you have just noticed for now there is no real support for the "stable" release, i.e. bugs are fixed in the 0.3 milestone release but not in the "stable" 0.2. - A "stable" release should offer backwards compatibility. There were some very incompatible changes from 0.2 to 0.3 (with the change in the package name being the most visible one). In my opinion both would not be acceptable for a 1.x release. > It may get more use if it's not thought of as "Alpha" software. You might be right, though I notice that there is generally a very small Java community around Asterisk and we probably won't be able to convince any of the "script-kiddies" (with script being PHP/Perl/... ;) to switch to Java even with the greatest Asterisk support. =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... |
|
From: Chris H. <ch...@as...> - 2006-08-29 20:31:17
|
Stefan Reuter wrote: > Hi Chris, > > Chris Howard wrote: > >> I'm having an issue with the DefaultAsteriskManager.originateCall not >> returning the channel after originate. I see that this issue is >> resolved in 0.3 per the bug report http://jira.reucon.org/browse/AJ-17 >> however I really need this for 0.2. Has anybody backported this? >> > > I am not aware of any backport. > What are the requirements that prevent you from switching to 0.3 > None (anymore). I migrated to 0.3 with very little problem. This release seems to be solid. On a side note, why are you sticking to the 0.X numbering scheme. I think that this should really be 1.X by now. It may get more use if it's not thought of as "Alpha" software. -- Chris Howard Email: ch...@as... Director Software Development Direct: 256.705.0262 Asteria Solutions Group, Inc. Main: 256.705.0277 2904 WestCorp BLVD, Suite 203 Fax: 256.705.0280 Huntsville, AL, 35805 Fax2Mail: 256.705.0262 http://www.asteriasgi.com Toll-Free: 877.ASGI.4.ME |
|
From: Stefan R. <sr...@re...> - 2006-08-29 20:16:43
|
Hi Chris, Chris Howard wrote: > I'm having an issue with the DefaultAsteriskManager.originateCall not=20 > returning the channel after originate. I see that this issue is=20 > resolved in 0.3 per the bug report http://jira.reucon.org/browse/AJ-17= =20 > however I really need this for 0.2. Has anybody backported this? I am not aware of any backport. What are the requirements that prevent you from switching to 0.3? =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... |
|
From: Wirsansky, E. <eya...@ve...> - 2006-08-29 20:08:21
|
Hello, I wanted to find out if anybody has implemented support for multiple languages in Asterisk while using Asterisk-Java. By Multilanguage support I mean being able to set the active language per call, upon which all file-based prompts for that call will be taken from a directory assigned to the active language, and all special-syntax prompts (such as numbers and dates) will use a syntax unique to the active language. Currently, my impression is that the inherent Multilanguage support of Asterisk and AGI is lacking, and needs to be augmented by Java programming on the application side, i.e. change the active directory by prepending its name to all prompt names, and use explicit rules for translating numbers and dates into a series of file-based prompts. Please let me know if you have any comments or suggestions. =20 Thanks, Eyal Wirsansky=20 |
|
From: Chris H. <ch...@as...> - 2006-08-29 12:44:50
|
I'm having an issue with the DefaultAsteriskManager.originateCall not returning the channel after originate. I see that this issue is resolved in 0.3 per the bug report http://jira.reucon.org/browse/AJ-17 however I really need this for 0.2. Has anybody backported this? -- Chris Howard Email: ch...@as... Director Software Development Direct: 256.705.0262 Asteria Solutions Group, Inc. Main: 256.705.0277 2904 WestCorp BLVD, Suite 203 Fax: 256.705.0280 Huntsville, AL, 35805 Fax2Mail: 256.705.0262 http://www.asteriasgi.com Toll-Free: 877.ASGI.4.ME |
|
From: Stefan R. <sr...@re...> - 2006-08-28 19:47:50
|
Karien Du Preez wrote: > Hi. >=20 > Is there any way to generate DTMF tones using asterisk-java? You can use the PlayDtmfAction (http://asterisk-java.org/0.3-m1/apidocs/org/asteriskjava/manager/action/= PlayDtmfAction.html) if you are using Asterisk-Java 0.3-m1 and Asterisk >=3D1.2.8 =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... |
|
From: Karien Du P. <kar...@gm...> - 2006-08-28 09:16:17
|
Hi. Is there any way to generate DTMF tones using asterisk-java? I have the following scenario: I originate a call using Originate, sometimes the number to dial is behind a private PBX routing extensions based on DTMF tones. Example the office number is 1234567 and the person I want to dial is on extension 33, I need to dial 1234567 and then generate DTMF tones for the extension. I know there is a asterisk application called SendDTMF that can be executed on a channel, but Im needing a sort of Manager command for this. Any help will be appreciated. Thanks. Regards. PS: Just some clarification, on the scenario. The Originate action dials the person (123467, extension 33) into an application, the MeetMe application. Thus there is only one side with a phone that can type in DTMF tones, but that channel is the one that needs to be routed to the appropiate extension using DTMF tones. This is why I need asterisk-java to generate the DTMF tones on the channel (resulting from the Originate action into the meetme room) for me. Regards. |
|
From: Karien Du P. <kar...@gm...> - 2006-08-28 08:56:01
|
Hi. Is there any way to generate DTMF tones using asterisk-java? I have the following scenario: I originate a call using Originate, sometimes the number to dial is behind a private PBX routing extensions based on DTMF tones. Example the office number is 1234567 and the person I want to dial is on extension 33, I need to dial 1234567 and then generate DTMF tones for the extension. I know there is a asterisk application called SendDTMF that can be executed on a channel, but Im needing a sort of Manager command for this. Any help will be appreciated. Thanks. Regards. |
|
From: Stefan R. <sr...@re...> - 2006-08-24 18:36:47
|
Juan Carlos Castellon wrote: > Here is my throuble, i need to listen all the Asterisk events. I > thought to use manager for this. I'm using Tomcat to host the > Asterisk-java framework and i need to start to listen the events once > the Tomcat server had been started. And do not wait for an invocation. implement ServletContextListener and register it in web.xml with the <listener-class> element - or use the Spring Framework. =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... |
|
From: Juan C. C. <jua...@gl...> - 2006-08-24 16:21:15
|
Hi everyone,
Here is my throuble, i need to listen all the Asterisk events. I thought
to use manager for this. I'm using Tomcat to host the Asterisk-java
framework and i need to start to listen the events once the Tomcat server
had been started. And do not wait for an invocation.
Have anybody solve this? or thinks how to solve it?
Thank you in advance,
Juan
|
|
From: Paul W. [Cellusys] <pau...@ce...> - 2006-08-22 14:25:24
|
http://maven.reucon.com/public/org/asteriskjava/asterisk-java/ Paul Dov Bigio wrote: > Where can I download it? > > []s > Dov > > -----Original Message----- > From: ast...@li... > [mailto:ast...@li...] On Behalf Of > Stefan Reuter > Sent: terça-feira, 22 de agosto de 2006 10:55 > To: ast...@li... > Subject: Re: [Asterisk-java-users] strange behaviour of managers connected > > Karien Du Preez wrote: >> It usually happens, when you stop asterisk while applications are >> connected via the Manager API to it. Once you start asterisk again, >> the application will try and restore the manager connection, but it >> seems as if something goes wrong with this reconnect. Resulting in the >> list of duplicate IPs and no manager account name when executing the >> 'show manager connected' command. > > This might be the known reconnect problem in Asterisk-Java 0.3. > Would you mind updating to 0.3-m1? > > =Stefan > > -- > reuter network consulting > Neusser Str. 110 > 50760 Koeln > Germany > Telefon: +49 221 1305699-0 > Telefax: +49 221 1305699-90 > E-Mail: sr...@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 > > ____________________________________________________________ > This inbound email to Cellusys.com has been scanned > by MessageLabs Email Security System for Cellusys. > www.messagelabs.com/email > ____________________________________________________________ ____________________________________________________________ This outbound email from Cellusys.com has been scanned by MessageLabs Email Security System for Cellusys. www.messagelabs.com/email ____________________________________________________________ |
|
From: Dov B. <dov...@gm...> - 2006-08-22 14:20:26
|
Where can I download it? []s Dov=20 -----Original Message----- From: ast...@li... [mailto:ast...@li...] On Behalf Of Stefan Reuter Sent: ter=E7a-feira, 22 de agosto de 2006 10:55 To: ast...@li... Subject: Re: [Asterisk-java-users] strange behaviour of managers = connected Karien Du Preez wrote: > It usually happens, when you stop asterisk while applications are=20 > connected via the Manager API to it. Once you start asterisk again,=20 > the application will try and restore the manager connection, but it=20 > seems as if something goes wrong with this reconnect. Resulting in the = > list of duplicate IPs and no manager account name when executing the=20 > 'show manager connected' command. This might be the known reconnect problem in Asterisk-Java 0.3. Would you mind updating to 0.3-m1? =3DStefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... |
|
From: Stefan R. <sr...@re...> - 2006-08-22 13:54:56
|
Karien Du Preez wrote: > It usually happens, when you stop asterisk while applications are > connected via the Manager API to it. Once you start asterisk again, the= > application will try and restore the manager connection, but it seems a= s > if something goes wrong with this reconnect. Resulting in the list of > duplicate IPs and no manager account name when executing the 'show > manager connected' command. This might be the known reconnect problem in Asterisk-Java 0.3. Would you mind updating to 0.3-m1? =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... |
|
From: Karien Du P. <kar...@gm...> - 2006-08-22 06:05:05
|
Hi. I have experienced similar behaviour. It usually happens, when you stop asterisk while applications are connected via the Manager API to it. Once you start asterisk again, the application will try and restore the manager connection, but it seems as if something goes wrong with this reconnect. Resulting in the list of duplicate IPs and no manager account name when executing the 'show manager connected' command. Hope this helps in tracking and fixing the problem. On 8/21/06, Dov Bigio <dov...@gm...> wrote: > > I will... As soon as I can reproduce the error! > > Thank you > Dov > > -----Original Message----- > From: ast...@li... > [mailto:ast...@li...] On Behalf Of > Stefan Reuter > Sent: sexta-feira, 18 de agosto de 2006 17:27 > To: ast...@li... > Subject: Re: [Asterisk-java-users] strange behaviour of managers connected > > Hi Dov, > > Dov Bigio wrote: > > Thank you for your help! I can try to send more details in case you > > find it necessary! > > please post the log messages (log4j with at least INFO, better DEBUG > level) of the client causing this issue. > Did the causing client experience any network difficulties before, e.g. > a disconnection from the asterisk server? > > Thanks > > =Stefan > > -- > reuter network consulting > Neusser Str. 110 > 50760 Koeln > Germany > Telefon: +49 221 1305699-0 > Telefax: +49 221 1305699-90 > E-Mail: sr...@re... > Jabber: sr...@ja... > > > > ------------------------------------------------------------------------- > 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: Dov B. <dov...@gm...> - 2006-08-21 14:42:02
|
I will... As soon as I can reproduce the error! Thank you Dov -----Original Message----- From: ast...@li... [mailto:ast...@li...] On Behalf Of Stefan Reuter Sent: sexta-feira, 18 de agosto de 2006 17:27 To: ast...@li... Subject: Re: [Asterisk-java-users] strange behaviour of managers connected Hi Dov, Dov Bigio wrote: > Thank you for your help! I can try to send more details in case you > find it necessary! please post the log messages (log4j with at least INFO, better DEBUG level) of the client causing this issue. Did the causing client experience any network difficulties before, e.g. a disconnection from the asterisk server? Thanks =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... |
|
From: Stefan R. <sr...@re...> - 2006-08-18 20:26:56
|
Hi Dov, Dov Bigio wrote: > Thank you for your help! I can try to send more details in case you fin= d > it necessary! please post the log messages (log4j with at least INFO, better DEBUG level) of the client causing this issue. Did the causing client experience any network difficulties before, e.g. a disconnection from the asterisk server? Thanks =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... |
|
From: Dov B. <dov...@gm...> - 2006-08-18 20:23:35
|
Hi Stephan, all,
I am using asterisk-java 0.2 for a Call Center monitoring application, where
I list all busy agents, and when the call cente manager clicks on the agent
name, a call to ChanSpy is made through the OriginateAction from
asterisk-java.
Everything works fine most of the time.... but sometimes I have the
following error condition on Asterisk:
It displays continuosly lots of warnings: WARNING[11737]: manager.c:1535
manager_event: Disconnecting slow (or gone) manager session!
And when this happens, if I run a show managers connected, I get the
following, with hundreds of times the same IP and no username...
(the IP that appears is from a client that connects to Asterisk through
asterisk-java....)
lv08*CLI> show manager connected
Username IP Address
200.234.206.109
200.234.206.109
200.234.206.109
200.234.206.109
200.234.206.109
200.234.206.109
200.234.206.109
200.234.206.109
.......
I have around 10 clients connecting to the manager port through the same
application, and each time this problem occurs with one client...
The only way we had to disconnects all this connections was to stop not only
the monitoring tool in the clients machines, but also restarting their JVM.
Thank you for your help! I can try to send more details in case you find it
necessary!
Dov
|
|
From: Vadim B. <Va...@nb...> - 2006-08-17 19:48:18
|
Try using the same -cp parameter for java as for javac
=20
________________________________
From: ast...@li...
[mailto:ast...@li...] On Behalf Of
Luis Perez
Sent: Thursday, August 17, 2006 3:46 PM
To: ast...@li...
Subject: [Asterisk-java-users] New in the list
=20
Hello, I'm new in the list, I don't know if this is the correct list.
I'm trying to exec a program, the example program of HelloEvents.
I have a problem that I think that it's a java problem, but maybe
someone know how to resolve it:=20
I compile the program like that:
javac -cp asterisk-java-0.2.jar HelloEvents.java
and then I exec the program:
#java HelloEvents
Exception in thread "main" java.lang.NoClassDefFoundError:
net/sf/asterisk/manager/ManagerEventHandler=20
at java.lang.ClassLoader.defineClass1(Native Method)
at java.lang.ClassLoader.defineClass(ClassLoader.java:620)
at
java.security.SecureClassLoader.defineClass(SecureClassLoader.java:124)
at java.net.URLClassLoader.defineClass(URLClassLoader.java:260)
at java.net.URLClassLoader.access$100(URLClassLoader.java:56)
at java.net.URLClassLoader$1.run(URLClassLoader.java:195)
at java.security.AccessController.doPrivileged(Native Method)
at java.net.URLClassLoader.findClass(URLClassLoader.java:188)
at java.lang.ClassLoader.loadClass(ClassLoader.java:306)
at sun.misc.Launcher$AppClassLoader.loadClass
(Launcher.java:268)
at java.lang.ClassLoader.loadClass(ClassLoader.java:251)
at java.lang.ClassLoader.loadClassInternal(ClassLoader.java:319)
I have this error.
--=20
Saludos,
Luis Perez Melia=20
|
|
From: Luis P. <f1....@gm...> - 2006-08-17 19:45:42
|
Hello, I'm new in the list, I don't know if this is the correct list.
I'm trying to exec a program, the example program of HelloEvents.
I have a problem that I think that it's a java problem, but maybe someone
know how to resolve it:
I compile the program like that:
javac -cp asterisk-java-0.2.jar HelloEvents.java
and then I exec the program:
#java HelloEvents
Exception in thread "main" java.lang.NoClassDefFoundError:
net/sf/asterisk/manager/ManagerEventHandler
at java.lang.ClassLoader.defineClass1(Native Method)
at java.lang.ClassLoader.defineClass(ClassLoader.java:620)
at java.security.SecureClassLoader.defineClass(
SecureClassLoader.java:124)
at java.net.URLClassLoader.defineClass(URLClassLoader.java:260)
at java.net.URLClassLoader.access$100(URLClassLoader.java:56)
at java.net.URLClassLoader$1.run(URLClassLoader.java:195)
at java.security.AccessController.doPrivileged(Native Method)
at java.net.URLClassLoader.findClass(URLClassLoader.java:188)
at java.lang.ClassLoader.loadClass(ClassLoader.java:306)
at sun.misc.Launcher$AppClassLoader.loadClass(Launcher.java:268)
at java.lang.ClassLoader.loadClass(ClassLoader.java:251)
at java.lang.ClassLoader.loadClassInternal(ClassLoader.java:319)
I have this error.
--
Saludos,
Luis Perez Melia
|
|
From: Allan K. <kam...@ya...> - 2006-08-15 11:08:25
|
Sergei, thanks for your input, I will now look into real-time asterisk. Allan. --- Sergei Polzunov <tr...@xb...> wrote: > Hi > > maybe it will be useful for you to use DB and > real-time asterisk > mechanism - you can create SIP channel at runtime > only by creating > record in the table in your DB. > > > Allan Kamau wrote: > > Hi all, > > I am unsuccessful in creating non-pre-existing SIP > > channels from AGI so as to make calls through > them. > > I am trying to make calls via AGI using > asterisk-java, > > and I have started by using the ManagerAPI and the > > sample code > > (http://asterisk-java.org/latest/tutorial.html) > but I > > can only make calls using channels that I have > already > > configured in the sip.conf and that have phones > > registered to them. > > Is there some class or method by which I can > create > > channels dynamically (at run time) using > > asterisk-java? > > > > Regards, > > Allan. > > > -- > Best regards, > Sergey Polzunov > [email: tr...@xb...] [ICQ: > #204206246] > [mood: ] > > > ------------------------------------------------------------------------- > 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 > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com |
|
From: Sergei P. <tr...@xb...> - 2006-08-14 20:14:14
|
Hi maybe it will be useful for you to use DB and real-time asterisk mechanism - you can create SIP channel at runtime only by creating record in the table in your DB. Allan Kamau wrote: > Hi all, > I am unsuccessful in creating non-pre-existing SIP > channels from AGI so as to make calls through them. > I am trying to make calls via AGI using asterisk-java, > and I have started by using the ManagerAPI and the > sample code > (http://asterisk-java.org/latest/tutorial.html) but I > can only make calls using channels that I have already > configured in the sip.conf and that have phones > registered to them. > Is there some class or method by which I can create > channels dynamically (at run time) using > asterisk-java? > > Regards, > Allan. -- Best regards, Sergey Polzunov [email: tr...@xb...] [ICQ: #204206246] [mood: ] |
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