Thread: Re: [Asterisk-java-users] Generating DTMF tones.
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From: Karien Du P. <kar...@gm...> - 2006-08-28 09:16:17
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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. |
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From: Stefan R. <sr...@re...> - 2006-08-28 19:47:50
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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... |
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From: Chris H. <ch...@as...> - 2006-08-29 12:44:50
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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 |
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From: Stefan R. <sr...@re...> - 2006-08-29 20:16:43
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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... |
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From: Chris H. <ch...@as...> - 2006-08-29 20:31:17
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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 |
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From: Stefan R. <sr...@re...> - 2006-08-29 21:03:09
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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... |
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From: Thameem A. <tha...@ya...> - 2006-08-30 02:31:24
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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. |
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From: Chris H. <ch...@as...> - 2006-08-30 04:35:25
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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 |
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From: Chris H. <ch...@as...> - 2006-08-30 05:04:18
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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);
}
------------------------------------------------------------------------
-------------------------- |
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From: Stefan R. <sr...@re...> - 2006-08-30 08:26:49
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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... |
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