asterisk-java-users — Asterisk-Java User List

Re: [Asterisk-java-users] Problems with PlayDtmfAction and Asterisk1.4.20.1

[E. M. <mar...@gm...>]

Maybe the problem is the DTMF signalling; Are you using SIP? Have you
checked the dtmf configuration in both asterisk sip.conf and the phone?

On Fri, Jun 6, 2008 at 5:31 PM, Martin Smith <ma...@be...> wrote:

>  Hello,
>
> It sounds like an Asterisk issue if Asterisk reports success but doesn't
> play the tone.
>
> Martin Smith, Systems Developer
> ma...@be...
> Bureau of Economic and Business Research
> University of Florida
> (352) 392-0171 Ext. 221
>
>
>  ------------------------------
> *From:* ast...@li... [mailto:
> ast...@li...] *On Behalf Of *Ing.
> Alvaro I. Parres Peredo
> *Sent:* Friday, June 06, 2008 11:13 AM
> *To:* ast...@li...
> *Subject:* [Asterisk-java-users] Problems with PlayDtmfAction and
> Asterisk1.4.20.1
>
>  Hi list, i'm trying to use the PlayDtmfAction via Manager… But is not
> working, any idea of a bug in this feature??
>
>
>
> When I send the action, I recive a response of Success but it's never play
> the tone on the phone.
>
>
>
> Thanks.
>
>
>
>
>
> Alvaro I. Parres Peredo
>
>  Director de IT
>
>  Grupo Xmarts SA de CV
>
>  Tel: +52 (33) 47 77 0110 Ext. 112
>
>           +52 (55) 47 77 3120
>
>           +52 (81) 12 47 6120
>
>           01 800  087 2260
>
>  Cel: +52 (33) 33 68 1087
>
>  alv...@xm...
>
>
>
>
> -------------------------------------------------------------------------
> Check out the new SourceForge.net Marketplace.
> It's the best place to buy or sell services for
> just about anything Open Source.
> http://sourceforge.net/services/buy/index.php
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] using originateToExtensionAsync, addPropertyChangeListener and a "Local/" channel

[Richard H. <ri...@ha...>]

http://pastebin.ca/1040549

Incase you want to see the clearer source


Richard

Richard Hamnett wrote:
> Hi there,
>
> I'm originating a call from a Local channel using 
> originateToExtensionAsync. I add a propertyChangeListener, for which i 
> want to get the CallDetailRecords for both legs of the call and output 
> the billableSeconds. However, with the current code below, the second 
> leg (we'll call this the company leg)'s CDR is always null.
>
> I would appreciate any help to try to figure out why it is not creating 
> a CDR for this event.
>
> Thanks in advance
>
> Richard
>
>
>
>     public static void main(String[] agrs) throws InterruptedException, 
> ManagerCommunicationException, NoSuchChannelException {
>
>         ManagerConnectionFactory mcf = new 
> ManagerConnectionFactory("ip","user","pass");
>         AsteriskServerImpl as = new 
> AsteriskServerImpl(mcf.createManagerConnection());
>         as.initialize();
>         OriginateCallback ocb = new OriginateCallbackTest(); 
>         as.originateToExtensionAsync("Local/3"+ clientNumber + 
> "@myContext/n", "myContext", compNumber, 1, 15000L, ocb);
>     }
>
>
>
>
>
> public class OriginateCallbackTest implements OriginateCallback{
>    
>     private AsteriskChannel clientChannel = null;
>     private AsteriskChannel compChannel= null;
>    
>
>     public void onBusy(AsteriskChannel arg0) {
>        
>     }
>
>     public void onDialing(AsteriskChannel arg0) {
>        
>     }
>
>     public void onFailure(LiveException arg0) {
>        
>     }
>
>     public void onNoAnswer(AsteriskChannel arg0) {
>        
>     }
>
>     public void onSuccess(AsteriskChannel chan) {
>        
>        
>         clientChannel = chan;
>        
>         chan.addPropertyChangeListener(new PropertyChangeListener(){
>
>             public void propertyChange(PropertyChangeEvent evt) {
>
>                
>                 if(evt.getNewValue()!=null)
>                     System.out.println("++++++++  THE VALUE NOW IS " + 
> evt.getNewValue().toString());
>                
>                 System.out.println("++++++++  THE PROPERTY NAME IS " + 
> evt.getPropertyName());               
>                
>                
>                 if(evt.getSource() instanceof AsteriskChannel) {
>                    
>                     /*
>                      * This is an event that happened on the clientChannel
>                      */
>                     if(evt.getSource() == clientChannel) {
>                    
>                         AsteriskChannel client = 
> (AsteriskChannel)evt.getSource();
>                         AsteriskChannel comp = client.getDialedChannel();
>                        
>                         CallDetailRecord clientCdr = 
> client.getCallDetailRecord();
>                         CallDetailRecord compCdr = 
> comp.getCallDetailRecord();
>                        
>                        
>                         if(clientCdr!=null){
>                             System.out.println("+++++++++++++ Client CDR 
> billable seconds is "  + clientCdr.getBillableSeconds());
>                         }else{
>                             System.out.println("+++++++++++++ CDR for 
> client was null");
>                         }
>                        
>                        
>                         if(comp !=null) {
>                             if(compCdr != null){
>                                 System.out.println("+++++++++++++ 
> Company CDR billable seconds is " + 
> comp.getCallDetailRecord().getBillableSeconds());
>                             }else{
>                                 System.out.println("+++++++++++++ CDR 
> for company was null");
>                             }
>                         }
>                     }
>
>
>                 }
>        
>                
>             }
>            
>         });
>        
>     }
>
> }
>
>
>
>
>
>
>
> -------------------------------------------------------------------------
> Check out the new SourceForge.net Marketplace.
> It's the best place to buy or sell services for
> just about anything Open Source.
> http://sourceforge.net/services/buy/index.php
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>   

[Asterisk-java-users] using originateToExtensionAsync, addPropertyChangeListener and a "Local/" channel

[Richard H. <ri...@ha...>]

Hi there,

I'm originating a call from a Local channel using 
originateToExtensionAsync. I add a propertyChangeListener, for which i 
want to get the CallDetailRecords for both legs of the call and output 
the billableSeconds. However, with the current code below, the second 
leg (we'll call this the company leg)'s CDR is always null.

I would appreciate any help to try to figure out why it is not creating 
a CDR for this event.

Thanks in advance

Richard



    public static void main(String[] agrs) throws InterruptedException, 
ManagerCommunicationException, NoSuchChannelException {

        ManagerConnectionFactory mcf = new 
ManagerConnectionFactory("ip","user","pass");
        AsteriskServerImpl as = new 
AsteriskServerImpl(mcf.createManagerConnection());
        as.initialize();
        OriginateCallback ocb = new OriginateCallbackTest(); 
        as.originateToExtensionAsync("Local/3"+ clientNumber + 
"@myContext/n", "myContext", compNumber, 1, 15000L, ocb);
    }





public class OriginateCallbackTest implements OriginateCallback{
   
    private AsteriskChannel clientChannel = null;
    private AsteriskChannel compChannel= null;
   

    public void onBusy(AsteriskChannel arg0) {
       
    }

    public void onDialing(AsteriskChannel arg0) {
       
    }

    public void onFailure(LiveException arg0) {
       
    }

    public void onNoAnswer(AsteriskChannel arg0) {
       
    }

    public void onSuccess(AsteriskChannel chan) {
       
       
        clientChannel = chan;
       
        chan.addPropertyChangeListener(new PropertyChangeListener(){

            public void propertyChange(PropertyChangeEvent evt) {

               
                if(evt.getNewValue()!=null)
                    System.out.println("++++++++  THE VALUE NOW IS " + 
evt.getNewValue().toString());
               
                System.out.println("++++++++  THE PROPERTY NAME IS " + 
evt.getPropertyName());               
               
               
                if(evt.getSource() instanceof AsteriskChannel) {
                   
                    /*
                     * This is an event that happened on the clientChannel
                     */
                    if(evt.getSource() == clientChannel) {
                   
                        AsteriskChannel client = 
(AsteriskChannel)evt.getSource();
                        AsteriskChannel comp = client.getDialedChannel();
                       
                        CallDetailRecord clientCdr = 
client.getCallDetailRecord();
                        CallDetailRecord compCdr = 
comp.getCallDetailRecord();
                       
                       
                        if(clientCdr!=null){
                            System.out.println("+++++++++++++ Client CDR 
billable seconds is "  + clientCdr.getBillableSeconds());
                        }else{
                            System.out.println("+++++++++++++ CDR for 
client was null");
                        }
                       
                       
                        if(comp !=null) {
                            if(compCdr != null){
                                System.out.println("+++++++++++++ 
Company CDR billable seconds is " + 
comp.getCallDetailRecord().getBillableSeconds());
                            }else{
                                System.out.println("+++++++++++++ CDR 
for company was null");
                            }
                        }
                    }


                }
       
               
            }
           
        });
       
    }

}

Re: [Asterisk-java-users] Problems with PlayDtmfAction and Asterisk1.4.20.1

[Martin S. <ma...@be...>]

Hello,
 
It sounds like an Asterisk issue if Asterisk reports success but doesn't
play the tone.

Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 


________________________________

	From: ast...@li...
[mailto:ast...@li...] On Behalf Of
Ing. Alvaro I. Parres Peredo
	Sent: Friday, June 06, 2008 11:13 AM
	To: ast...@li...
	Subject: [Asterisk-java-users] Problems with PlayDtmfAction and
Asterisk1.4.20.1
	
	

	Hi list, i'm trying to use the PlayDtmfAction via Manager... But
is not working, any idea of a bug in this feature??

	 

	When I send the action, I recive a response of Success but it's
never play the tone on the phone.

	 

	Thanks.

	 

	 

	Alvaro I. Parres Peredo

	 Director de IT

	 Grupo Xmarts SA de CV

	 Tel: +52 (33) 47 77 0110 Ext. 112

	          +52 (55) 47 77 3120 

	          +52 (81) 12 47 6120 

	          01 800  087 2260 

	 Cel: +52 (33) 33 68 1087

	 alv...@xm...

	 

[Asterisk-java-users] Problems with PlayDtmfAction and Asterisk 1.4.20.1

[Ing. A. I. P. P. <ar...@xm...>]

Hi list, i'm trying to use the PlayDtmfAction via Manager. But is not
working, any idea of a bug in this feature??

 

When I send the action, I recive a response of Success but it's never play
the tone on the phone.

 

Thanks.

 

 

Alvaro I. Parres Peredo

 Director de IT

 Grupo Xmarts SA de CV

 Tel: +52 (33) 47 77 0110 Ext. 112

          +52 (55) 47 77 3120 

          +52 (81) 12 47 6120 

          01 800  087 2260 

 Cel: +52 (33) 33 68 1087

 alv...@xm...

 

Re: [Asterisk-java-users] unhandled events

[Martin S. <ma...@be...>]

Thanks for the reports. You haven't added too many events -- we
definitely need to handle them :)

Stefan appears to have already put in issues to JIRA for them, if you'd
like to track the status.

Cheers,

Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

> -----Original Message-----
> From: ast...@li... 
> [mailto:ast...@li...] On 
> Behalf Of Andy Burns
> Sent: Thursday, June 05, 2008 9:26 AM
> To: ast...@li...
> Subject: [Asterisk-java-users] unhandled events
> 
> I'm using asterisk-java-1.0.0-20080604.214555-170 and seeing several 
> unhandled events, such as
> 
> 
> 05-Jun-2008 14:11:04 
> org.asteriskjava.manager.internal.EventBuilderImpl 
> buildEvent
> INFO: No event class registered for event type 'varset', attributes: 
> {uniqueid=1212670244.39, event=VarSet, privilege=dialplan,all, 
> value=SUCCESS, variable=AGISTATUS, 
> channel=SIP/gradwell-out1-02495290}. 
> Please report at http://jira.reucon.org/browse/AJ
> 
> 
> 05-Jun-2008 14:09:57 
> org.asteriskjava.manager.internal.EventBuilderImpl 
> buildEvent
> INFO: No event class registered for event type 'rtcpsent', 
> attributes: 
> {dlsr=57662.8420 (sec), fractionlost=0, iajitter=0.0029, 
> cumulativeloss=0, sentrtp=18088, ourssrc=1551750496, theirlastsr=0, 
> event=RTCPSent, privilege=reporting,all, sentoctets=15840, 
> sentpackets=99, sentntp=1212670270.3452047360}. Please report at 
> http://jira.reucon.org/browse/AJ
> 
> 
> 05-Jun-2008 14:11:04 
> org.asteriskjava.manager.internal.EventBuilderImpl 
> buildEvent
> INFO: No event class registered for event type 'rtpreceiverstat', 
> attributes: {receivedpackets=4608, ssrc=240272389, rrcount=4, 
> lostpackets=0, event=RTPReceiverStat, privilege=reporting,all, 
> jitter=0.0013, transit=0.0000}. Please report at 
> http://jira.reucon.org/browse/AJ
> 
> 
> 05-Jun-2008 14:11:04 
> org.asteriskjava.manager.internal.EventBuilderImpl 
> buildEvent
> INFO: No event class registered for event type 
> 'usereventivrreply|callid: 1|userresponse: t', attributes: 
> {userevent=ivrReply|callId: 1|userResponse: T, event=UserEvent, 
> privilege=user,all}. Please report at http://jira.reucon.org/browse/AJ
> 
> 
> The last one corresponds to my custom userevent used to report call 
> status back to my main app, should the others be handled somehow, or 
> gracefully ignored? Or perhaps I've enabled too many events in 
> manager.conf with
> 
> read = 
> system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
> write = system,call,agent,user,config,command,reporting,originate
> 
> 
> 

[Asterisk-java-users] unhandled events

[Andy B. <and...@pr...>]

I'm using asterisk-java-1.0.0-20080604.214555-170 and seeing several 
unhandled events, such as


05-Jun-2008 14:11:04 org.asteriskjava.manager.internal.EventBuilderImpl 
buildEvent
INFO: No event class registered for event type 'varset', attributes: 
{uniqueid=1212670244.39, event=VarSet, privilege=dialplan,all, 
value=SUCCESS, variable=AGISTATUS, channel=SIP/gradwell-out1-02495290}. 
Please report at http://jira.reucon.org/browse/AJ


05-Jun-2008 14:09:57 org.asteriskjava.manager.internal.EventBuilderImpl 
buildEvent
INFO: No event class registered for event type 'rtcpsent', attributes: 
{dlsr=57662.8420 (sec), fractionlost=0, iajitter=0.0029, 
cumulativeloss=0, sentrtp=18088, ourssrc=1551750496, theirlastsr=0, 
event=RTCPSent, privilege=reporting,all, sentoctets=15840, 
sentpackets=99, sentntp=1212670270.3452047360}. Please report at 
http://jira.reucon.org/browse/AJ


05-Jun-2008 14:11:04 org.asteriskjava.manager.internal.EventBuilderImpl 
buildEvent
INFO: No event class registered for event type 'rtpreceiverstat', 
attributes: {receivedpackets=4608, ssrc=240272389, rrcount=4, 
lostpackets=0, event=RTPReceiverStat, privilege=reporting,all, 
jitter=0.0013, transit=0.0000}. Please report at 
http://jira.reucon.org/browse/AJ


05-Jun-2008 14:11:04 org.asteriskjava.manager.internal.EventBuilderImpl 
buildEvent
INFO: No event class registered for event type 
'usereventivrreply|callid: 1|userresponse: t', attributes: 
{userevent=ivrReply|callId: 1|userResponse: T, event=UserEvent, 
privilege=user,all}. Please report at http://jira.reucon.org/browse/AJ


The last one corresponds to my custom userevent used to report call 
status back to my main app, should the others be handled somehow, or 
gracefully ignored? Or perhaps I've enabled too many events in 
manager.conf with

read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
write = system,call,agent,user,config,command,reporting,originate

Re: [Asterisk-java-users] AsyncAgi

[Andy B. <and...@pr...>]

On 04/06/2008 23:12, Stefan Reuter wrote:
> Andy Burns wrote:
>> I'd like to convert to the new asyncAGI, can the AsyncAgiServer attach
>> its EventListener to my existing ManagerConnection, or does it require
>> another ManagerConnection?
> 
> It can use the same ManagerConnection.

thanks, I may need to try a slightly older "CVS" version, I'm seeing a 
few exceptions.

With fastagi, you could call various scripts from the dialplan with 
Agi(http://myserver/myscript.agi) and fastagi-mapping.properties,

With asycnagi it seems that from the dialplan Agi(agi:async) only 
provides for a single script, or have I misunderstood and perhaps I can 
somehow use Agi(myscript:async) to do mapping?

Re: [Asterisk-java-users] AsyncAgi

[Stefan R. <ste...@re...>]

Andy Burns wrote:
> I'd like to convert to the new asyncAGI, can the AsyncAgiServer attach
> its EventListener to my existing ManagerConnection, or does it require
> another ManagerConnection?

It can use the same ManagerConnection.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  ste...@re...
Jabber:  ste...@re...
WWW:     http://www.reucon.com

Steuernummern 215/5140/1791 USt-IdNr. DE220701760

[Asterisk-java-users] AsyncAgi

[Andy B. <and...@pr...>]

The main part of my app already uses a ManagerConnection and sends 
asynchronous OriginateActions, listening for the corresponding 
MangagerEvents and ManagerResponses.

The app currently delivers calls to my fastAGI script, running as a 
separate DefaultAgiServer, when finished, the script sends a UserEvent 
back to the main app.

I'd like to convert to the new asyncAGI, can the AsyncAgiServer attach 
its EventListener to my existing ManagerConnection, or does it require 
another ManagerConnection?

Re: [Asterisk-java-users] creating sip users and peers programmatically

[Martin S. <ma...@be...>]

Hello Gabriel,

It looks like there is no Manager interface or console way to create a
sip peer directly, though you could use the GetConfig and PutConfig
actions on sip.conf, and then issue a sip reload.

Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

> -----Original Message-----
> From: ast...@li... 
> [mailto:ast...@li...] On 
> Behalf Of Gabriel Bermudez
> Sent: Tuesday, June 03, 2008 7:05 PM
> To: Asterisk-Java User List
> Subject: [Asterisk-java-users] creating sip users and peers 
> programmatically
> 
> Hi list,
> 
> Is there a way to programmatically create a sip user or peer.  For 
> example, I can read the users data from a database and through the 
> manager configure asterisk with this users.  I first thought on using 
> the asterisk's realtime extension and modify the database but 
> it would 
> be better if there where another way.  Another thing, I know 
> this is not 
> the place for this but who good asterisk scales up with 
> realtime?  who 
> many users can be registered at the same time?
> 
> Thanks for your answers
> 
> Good bye
> 
> --------------------------------------------------------------
> -----------
> Check out the new SourceForge.net Marketplace.
> It's the best place to buy or sell services for
> just about anything Open Source.
> http://sourceforge.net/services/buy/index.php
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> 

[Asterisk-java-users] creating sip users and peers programmatically

[Gabriel B. <elg...@gm...>]

Hi list,

Is there a way to programmatically create a sip user or peer.  For 
example, I can read the users data from a database and through the 
manager configure asterisk with this users.  I first thought on using 
the asterisk's realtime extension and modify the database but it would 
be better if there where another way.  Another thing, I know this is not 
the place for this but who good asterisk scales up with realtime?  who 
many users can be registered at the same time?

Thanks for your answers

Good bye

Re: [Asterisk-java-users] Setting the accountcode when firing an originateToExtensionAsync

[Richard H. <ri...@ha...>]

Hi Martin


I tried the combination of "CDR(accountcode)" "account" "Account" 
"accountcode" and nothing seems to work unfortunately.




Rick






Hi Richard,

Have you tried using "Account" instead of "CDR(accountcode)"?

The spec for the Manager API's originate action calls it "Account" --
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
...

If that doesn't work, it would be trivial for us to add it to the live
API as a method argument, so you can call it more directly.

Let us know :)


Martin Smith, Systems Developer
martins@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221




________________________________

From: asterisk-java-users-bounces@li...
[mailto:asterisk-java-users-bounces@li...] On Behalf Of
Richard Hamnett
Sent: Tuesday, June 03, 2008 4:38 PM
To: asterisk-java-users@li...
Subject: [Asterisk-java-users] Setting the accountcode when
firing anoriginateToExtensionAsync


I was wondering if someone can help me, i'm using
originateToExtensionAsync to create a call but I cannot figure out a way
to set the account code within the method

I tried something along the lines of:


Map<String, String> m = new HashMap<String, String>();

m.put("CDR(accountcode)", accountCode);

serverBean.getServer().originateToExtensionAsync("Local/3"+
exten + "@cmyContext/n", "myContext", companyNumber, 1, 15000L, new
CallerId(companyName, companyNumber), m, callbackBean);



Can anyone kindly suggest how to do so?

Thanks,

Richard

Re: [Asterisk-java-users] Setting the accountcode when firing anoriginateToExtensionAsync

[Martin S. <ma...@be...>]

Hi Richard,
 
Have you tried using "Account" instead of "CDR(accountcode)"?
 
The spec for the Manager API's originate action calls it "Account" --
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
...
 
If that doesn't work, it would be trivial for us to add it to the live
API as a method argument, so you can call it more directly.
 
Let us know :)
 

Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 


________________________________

	From: ast...@li...
[mailto:ast...@li...] On Behalf Of
Richard Hamnett
	Sent: Tuesday, June 03, 2008 4:38 PM
	To: ast...@li...
	Subject: [Asterisk-java-users] Setting the accountcode when
firing anoriginateToExtensionAsync
	
	
	I was wondering if someone can help me, i'm using
originateToExtensionAsync to create a call but I cannot figure out a way
to set the account code within the method
	
	I tried something along the lines of:
	
	
	Map<String, String> m = new HashMap<String, String>();
	            
	m.put("CDR(accountcode)", accountCode);
	
	serverBean.getServer().originateToExtensionAsync("Local/3"+
exten + "@cmyContext/n", "myContext", companyNumber, 1, 15000L, new
CallerId(companyName, companyNumber), m, callbackBean);
	
	
	
	Can anyone kindly suggest how to do so?
	
	Thanks,
	
	Richard
	

[Asterisk-java-users] Setting the accountcode when firing an originateToExtensionAsync

[Richard H. <ri...@ha...>]

I was wondering if someone can help me, i'm using 
originateToExtensionAsync to create a call but I cannot figure out a way 
to set the account code within the method

I tried something along the lines of:


Map<String, String> m = new HashMap<String, String>();
           
m.put("CDR(accountcode)", accountCode);

serverBean.getServer().originateToExtensionAsync("Local/3"+ exten + 
"@cmyContext/n", "myContext", companyNumber, 1, 15000L, new 
CallerId(companyName, companyNumber), m, callbackBean);



Can anyone kindly suggest how to do so?

Thanks,

Richard

Re: [Asterisk-java-users] Asterisk-java-users Digest, Vol 26, Issue 1

[Martin S. <ma...@be...>]

Modified the JavaDoc of StreamFile to reflect that it follows
conventions and paths of dialplan apps like Playback or Background. :)
 

Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 


________________________________

	From: ast...@li...
[mailto:ast...@li...] On Behalf Of
Olivier
	Sent: Tuesday, June 03, 2008 8:04 AM
	To: ast...@li...
	Subject: Re: [Asterisk-java-users] Asterisk-java-users Digest,
Vol 26,Issue 1
	
	


		------------------------------
		
		Message: 5
		Date: Tue, 3 Jun 2008 11:27:18 +0200
		From: "Maciek Tokarski" <mlo...@gm...>
		Subject: Re: [Asterisk-java-users] My first FastAGI
script : document
		       "welcome"file content and location
		To: <ast...@li...>
		Message-ID: <10D408F862A444D7BB1504D416CAE1E6@mt>
		Content-Type: text/plain; charset="us-ascii"
		
		Hi
		
		"welcome" is a sound file (about sounds in asterisk see
		http://www.voip-info.org/wiki-Asterisk+sound+files) it
should be located in
		/var/lib/asterisk/sounds directory.
		
		You can produce it simply recording yourself and
converting file into proper
		format or you can use some example sounds files that
already should be in
		/var/lib/asterisk/sounds
		
		 Reagrds, Maciek


	Maybe, this could be added to online documentation ? 
	





		 _____
		
		From: ast...@li...
	
[mailto:ast...@li...] On Behalf Of
		Olivier
		Sent: Tuesday, June 03, 2008 11:19 AM
		To: ast...@li...
		Subject: [Asterisk-java-users] My first FastAGI script :
document
		"welcome"file content and location
		
		
		
		Hi,
		
		I carefully followed instructions here
		http://asterisk-java.org/0.3.1/tutorial.html when
writing my very first
		FastAGI script.
		I couldn't find anything on this "welcome" file which is
streamed to calling
		party in HelloAgiScript example :
		
		streamFile("welcome");
		
		
		
		1. Is "welcome" a file ?
		2. Where should it be located ?
		3. How can you produce it ?
		
		Anyway, congrats for writing such detailed instructions.
		
		Regards
		
		-------------- next part --------------
		An HTML attachment was scrubbed...
		
		-

Re: [Asterisk-java-users] Strange exception occurs when trying toaccess policy file

[Martin S. <ma...@be...>]

Hello Balaji,
 
You're running into security policies enforced by the JVM you're running
code inside. It hasn't anything to do with Asterisk-Java.
 
Try your one statement inside a simple main method:
System.getProperty("user.dir");
 
It should throw the same exception. Consult Google for ways of changing
the current security policy.
 

Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 


________________________________

	From: ast...@li...
[mailto:ast...@li...] On Behalf Of
Balaji Chinan
	Sent: Monday, June 02, 2008 4:54 AM
	To: ast...@li...
	Subject: [Asterisk-java-users] Strange exception occurs when
trying toaccess policy file
	
	

	Hi All,
	
	   1. I have a problem with my RMI code when I am trying to
connect one java application on Linux environment. When I run my rmi
code it gives the following error,
	java.security.AccessControlException: access denied
(java.util.PropertyPermission java.security.policy write)
	       at
java.security.AccessControlContext.checkPermission(AccessControlContext.
java:264)
	       at
java.security.AccessController.checkPermission(AccessController.java:427
)
	       at
java.lang.SecurityManager.checkPermission(SecurityManager.java:532)
	       at java.lang.System.setProperty(System.java:699)
	       at HelloAgiScript.service(HelloAgiScript.java:48)
	       at
org.asteriskjava.fastagi.internal.AgiConnectionHandler.runScript(AgiConn
ectionHandler.java:144)
	       at
org.asteriskjava.fastagi.internal.AgiConnectionHandler.run(AgiConnection
Handler.java:116)
	       at
java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecuto
r.java:650)
	       at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.ja
va:675)
	       at java.lang.Thread.run(Thread.java:595)
	
	
	I have setup the security manager in my code as
	System.setProperty("java.security.policy",
"HelloAgiScript.policy");
	       if (System.getSecurityManager() == null) {
	           System.setSecurityManager(new RMISecurityManager());
	       }
	
	And I have created the security policy file as 
	grant {
	    permission java.security.AllPermission;
	};
	
	Can anyone assist me to solve this problem please...
	
	2 . Also I got exception,
	java.security.AccessControlException: access denied
(java.util.PropertyPermission user.dir read)
	        at
java.security.AccessControlContext.checkPermission(AccessControlContext.
java:264)
	        at
java.security.AccessController.checkPermission(AccessController.java:427
)
	        at
java.lang.SecurityManager.checkPermission(SecurityManager.java:532)
	        at
java.lang.SecurityManager.checkPropertyAccess(SecurityManager.java:1285)
	        at java.lang.System.getProperty(System.java:628)
	        at HelloAgiScript.service(HelloAgiScript.java:51)
	        at
org.asteriskjava.fastagi.internal.AgiConnectionHandler.runScript(AgiConn
ectionHandler.java:144)
	        at
org.asteriskjava.fastagi.internal.AgiConnectionHandler.run(AgiConnection
Handler.java:116)
	        at
java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecuto
r.java:650)
	        at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.ja
va:675)
	        at java.lang.Thread.run(Thread.java:595)
	
	when I try to get the current path using the line "String
filePath = System.getProperty("user.dir");"..Please tell whether these
type of functionalities allowed in Asterisk-java environment.
	
	Thanks/Regards,
	 Sourab
	
	

Re: [Asterisk-java-users] Asterisk-java-users Digest, Vol 26, Issue 1

[Olivier <oza...@my...>]

> ------------------------------
>
> Message: 5
> Date: Tue, 3 Jun 2008 11:27:18 +0200
> From: "Maciek Tokarski" <mlo...@gm...>
> Subject: Re: [Asterisk-java-users] My first FastAGI script : document
>        "welcome"file content and location
> To: <ast...@li...>
> Message-ID: <10D408F862A444D7BB1504D416CAE1E6@mt>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi
>
> "welcome" is a sound file (about sounds in asterisk see
> http://www.voip-info.org/wiki-Asterisk+sound+files) it should be located
> in
> /var/lib/asterisk/sounds directory.
>
> You can produce it simply recording yourself and converting file into
> proper
> format or you can use some example sounds files that already should be in
> /var/lib/asterisk/sounds
>
>  Reagrds, Maciek


Maybe, this could be added to online documentation ?

>
>
>
>
>  _____
>
> From: ast...@li...
> [mailto:ast...@li...] On Behalf Of
> Olivier
> Sent: Tuesday, June 03, 2008 11:19 AM
> To: ast...@li...
> Subject: [Asterisk-java-users] My first FastAGI script : document
> "welcome"file content and location
>
>
>
> Hi,
>
> I carefully followed instructions here
> http://asterisk-java.org/0.3.1/tutorial.html when writing my very first
> FastAGI script.
> I couldn't find anything on this "welcome" file which is streamed to
> calling
> party in HelloAgiScript example :
>
> streamFile("welcome");
>
>
>
> 1. Is "welcome" a file ?
> 2. Where should it be located ?
> 3. How can you produce it ?
>
> Anyway, congrats for writing such detailed instructions.
>
> Regards
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
>
> -

Re: [Asterisk-java-users] My first FastAGI script : document "welcome" file content and location

[Carlos G M. <tr...@hu...>]

Sound file in /var/lib/asterisk/sounds...
Asterisk can take care of transcoding if needed.
-CArlos

Olivier @ 03/06/2008 06:19 -0300 dixit:
> Hi,
> 
> I carefully followed instructions here 
> http://asterisk-java.org/0.3.1/tutorial.html when writing my very first 
> FastAGI script.
> I couldn't find anything on this "welcome" file which is streamed to 
> calling party in HelloAgiScript example :
> 
> streamFile("welcome");
> 
> 
> 
> 1. Is "welcome" a file ?
> 2. Where should it be located ?
> 3. How can you produce it ?
> 
> Anyway, congrats for writing such detailed instructions.
> 
> Regards
> 
> 
> ------------------------------------------------------------------------
> 
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2008.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

-- 
Carlos G Mendioroz  <tr...@hu...>  LW7 EQI  Argentina

Re: [Asterisk-java-users] My first FastAGI script : document "welcome"file content and location

[Maciek T. <mlo...@gm...>]

Hi

"welcome" is a sound file (about sounds in asterisk see
http://www.voip-info.org/wiki-Asterisk+sound+files) it should be located in
/var/lib/asterisk/sounds directory.

You can produce it simply recording yourself and converting file into proper
format or you can use some example sounds files that already should be in
/var/lib/asterisk/sounds

 Reagrds, Maciek

 

  _____  

From: ast...@li...
[mailto:ast...@li...] On Behalf Of
Olivier
Sent: Tuesday, June 03, 2008 11:19 AM
To: ast...@li...
Subject: [Asterisk-java-users] My first FastAGI script : document
"welcome"file content and location

 

Hi,

I carefully followed instructions here
http://asterisk-java.org/0.3.1/tutorial.html when writing my very first
FastAGI script.
I couldn't find anything on this "welcome" file which is streamed to calling
party in HelloAgiScript example :

streamFile("welcome");



1. Is "welcome" a file ?
2. Where should it be located ?
3. How can you produce it ?

Anyway, congrats for writing such detailed instructions.

Regards

[Asterisk-java-users] My first FastAGI script : document "welcome" file content and location

[Olivier <oza...@my...>]

Hi,

I carefully followed instructions here
http://asterisk-java.org/0.3.1/tutorial.html when writing my very first
FastAGI script.
I couldn't find anything on this "welcome" file which is streamed to calling
party in HelloAgiScript example :

streamFile("welcome");



1. Is "welcome" a file ?
2. Where should it be located ?
3. How can you produce it ?

Anyway, congrats for writing such detailed instructions.

Regards

[Asterisk-java-users] Strange exception occurs when trying to access policy file

[Balaji C. <bal...@gm...>]

Hi All,
   1. I have a problem with my RMI code when I am trying to connect one java
application on Linux environment. When I run my rmi code it gives the
following error,
java.security.AccessControlException: access denied
(java.util.PropertyPermission java.security.policy write)
       at
java.security.AccessControlContext.checkPermission(AccessControlContext.java:264)
       at
java.security.AccessController.checkPermission(AccessController.java:427)
       at
java.lang.SecurityManager.checkPermission(SecurityManager.java:532)
       at java.lang.System.setProperty(System.java:699)
       at HelloAgiScript.service(HelloAgiScript.java:48)
       at
org.asteriskjava.fastagi.internal.AgiConnectionHandler.runScript(AgiConnectionHandler.java:144)
       at
org.asteriskjava.fastagi.internal.AgiConnectionHandler.run(AgiConnectionHandler.java:116)
       at
java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:650)
       at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:675)
       at java.lang.Thread.run(Thread.java:595)


I have setup the security manager in my code as
System.setProperty("java.security.policy", "HelloAgiScript.policy");
       if (System.getSecurityManager() == null) {
           System.setSecurityManager(new RMISecurityManager());
       }

And I have created the security policy file as
grant {
    permission java.security.AllPermission;
};

Can anyone assist me to solve this problem please...

2 . Also I got exception,
java.security.AccessControlException: access denied
(java.util.PropertyPermission user.dir read)
        at
java.security.AccessControlContext.checkPermission(AccessControlContext.java:264)
        at
java.security.AccessController.checkPermission(AccessController.java:427)
        at
java.lang.SecurityManager.checkPermission(SecurityManager.java:532)
        at
java.lang.SecurityManager.checkPropertyAccess(SecurityManager.java:1285)
        at java.lang.System.getProperty(System.java:628)
        at HelloAgiScript.service(HelloAgiScript.java:51)
        at
org.asteriskjava.fastagi.internal.AgiConnectionHandler.runScript(AgiConnectionHandler.java:144)
        at
org.asteriskjava.fastagi.internal.AgiConnectionHandler.run(AgiConnectionHandler.java:116)
        at
java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:650)
        at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:675)
        at java.lang.Thread.run(Thread.java:595)

when I try to get the current path using the line "String filePath =
System.getProperty("user.dir");"..Please tell whether these type of
functionalities allowed in Asterisk-java environment.

Thanks/Regards,
 Sourab

Re: [Asterisk-java-users] Missing events through the manager

[Stefan R. <ste...@re...>]

Hi,

> As a workaround, could we modify the live api to compare the current
> callerid recrded in the channel with the callerid received in a newstate
> event, and update the channel if necessary (thus firing a PCE on the
> channel too), or even generate and handle a virtual "newcallerid" even ?
> Any clue ?

Yes I think we should (if we don't do that already) update the caller id
if we receive one with a NewStateEvent. We only have to make sure that
we don't overwrite it with null for older versions of Asterisk where the
property might not be available.

Can you check that this works and provide a patch in Jira?

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  ste...@re...
Jabber:  ste...@re...
WWW:     http://www.reucon.com

Steuernummern 215/5140/1791 USt-IdNr. DE220701760

[Asterisk-java-users] Missing events through the manager

[Gaëtan M. <gm...@ea...>]

Hi Stefan

I'm running across a problem I never noticed before with asterisk. I  
think it worked as expected with asterisk 1.2 but seems to have  
changed somewhere between 1.4 and 1.4.11.

The problem I'm facing is related to outgoing legs from queues, where  
the caller ID is (now ?) missing in the live API's channel object.  
This is most probably a problem on the asterisk side (although as the  
manager api is not officially documented  I can't really blame them).

When a leg (with a correct caller id) arrives in a queue, except usual  
queue events, here is what I see on the manager:

1/ One event for a new channel to the queue member, sate down and  
caller id unknown

Event: Newchannel.
Privilege: call,all.
Channel: SIP/1002-09829228.
State: Down.
CallerIDNum: <unknown>.
CallerIDName: <unknown>.
Uniqueid: 1211559337.1.


2/  NewState event when the outboud leg starts to ring

Event: Newstate.
Privilege: call,all.
Channel: SIP/1002-09829228.
State: Ringing.
CallerID: 1001.
CallerIDName: TESTQUEUE:Gaetan Minet.
Uniqueid: 1211559337.1.

(Following events are useless).


So there is no newCallerId event and as a consequence the live api  
never updates the channel information.

In the past (pre-1.4) , iirc, we got a newcallerid event between (1)  
and (2).

I don't know if the real problem is
- the "unknown" in (1) (Maybe asterisk sends the event before the  
callerid is set on the channel ??) ,
- the missing newcallerid event between (1) and (2) that I thing I  
received with 1.2.x
- or simply maybe the newstate event setting the caller id is the  
expected behavior for recent asterisk.

Has someone already encountered this ?

As a workaround, could we modify the live api to compare the current  
callerid recrded in the channel with the callerid received in a  
newstate event, and update the channel if necessary (thus firing a PCE  
on the channel too), or even generate and handle a virtual  
"newcallerid" even ?
Any clue ?

Thanks

Kind regards

Gaetan



NB: Just saw this in the ChannelManager.java, maybe you already had to  
face a similar problem ? Maybe just ignoring the event (1) could also  
fix the problem then ?

AsteriskChannelImpl channel = getChannelImplById(event.getUniqueId());

         if (channel == null)
         {
             // NewStateEvent can occur instead of a NewChannelEvent
             channel = addNewChannel(
                     event.getUniqueId(), event.getChannel(),  
event.getDateReceived(),
                     event.getCallerIdNum(), event.getCallerIdName(),
                     ChannelState.valueOf(event.getChannelState()),  
null /* account code not available */);
         }

Re: [Asterisk-java-users] RES: RES: cdr events on live api

[Gaëtan M. <gm...@ea...>]

Hi,

As far as I know, Asterisk doesn't allow one to park a call through  
ami or agi. Or at least I never found any "documented" way of doing  
this.

However here is a simple workaround we use in our applications: we  
redirect the caller's leg (see ChannelImpl.redirect() in live api)  to  
the park extension (see in features.conf), or any dialplan extension  
that calls the Park() application in fact.

You will then receive a manager event telling at which parking  
position the caller can be retrieved (so later you just need to  
originate from the operator to that parking extension). Just match the  
channel name in your code.
If you use the live api, you will receive a PCE on the PARKED_AT  
property of the parked channel iirc.

NB:
- The problem however is that the caller will hear the parking  
position (because the "function" in res_features.so is really written  
for internal use using inband feature codes, and as such expects a  
bridged channel).
I think you could work around this if you set the channel's language  
variable to some unknown language (or an exiting language whose sound  
files are empty...). On my side I created a simple and trivial  
ParkSilent() asterisk application that I use instead of Park(), it's  
eventually easier.
- This will probably completely screw your cdr records, as any  
transfer in asterisk :(


Kind regards

Geatan

On 21/05/2008, at 20:38, Leonardo Lira wrote:

> Hi Martin,
>
> Let me try to explain to you what i am expecting from the hold command
> Suppose the next scenary:
> - A is a call center employer
> - B is a customer receiving an incoming call from A
> - During the call, A decides to check some information about B and  
> he tells
> B to wait for a while
> - In fact, what A does is to put the call in hold while he realize  
> some
> action and B listens to some configured music
> - After that A retrieve the call and start talking to B again
>
> Put a call in a queue or in a park will turn off the A channel, what  
> means
> that when A desires to retrieve the call his phone will ring again.
>
> Therefore, what i want to know is if there is some way to hold a  
> call just
> like the soft-phones do (X-lite for example), like:
> putChannelOnHold(channel), retrieveChannel(channel)
>
> I search the entire Java-asterisk API and found nothing similar to  
> that.
>
> Below is my extensions.conf
>
> Best regards
>
> ----------------------------------------------------------------------------
> -------------------------------
> ;!
> ;! Automatically generated configuration file
> ;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
> ;! Generator: Manager
> ;! Creation Date: Fri May 16 03:51:37 2008
> ;!
> [general]
> static = yes
> writeprotect = no
> autofallthrough = yes
> clearglobalvars = no
> priorityjumping = no
>
> [globals]
> trunk_1 = SIP/trunk_1
> trunk_2 = SIP/trunk_2
> trunk_2_cid = unknown
> trunk_1_cid = 090909
>
> [dundi-e164-canonical]
>
> [dundi-e164-customers]
>
> [dundi-e164-via-pstn]
>
> [dundi-e164-local]
> include => dundi-e164-canonical
> include => dundi-e164-customers
> include => dundi-e164-via-pstn
>
> [dundi-e164-switch]
> switch => DUNDi/e164
>
> [dundi-e164-lookup]
> include => dundi-e164-local
> include => dundi-e164-switch
>
> [macro-dundi-e164]
> exten => s,1,Goto(${ARG1},1)
> include => dundi-e164-lookup
>
> [iaxtel700]
> exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN: 
> 1}@iaxtel)
>
> [iaxprovider]
>
> [trunkint]
> exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
> exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [trunkld]
> exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
> exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [trunklocal]
> exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [trunktollfree]
> exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [international]
> ignorepat => 9
> include => longdistance
> include => trunkint
>
> [longdistance]
> ignorepat => 9
> include => local
> include => trunkld
>
> [local]
> ignorepat => 9
> include => default
> include => parkedcalls
> include => trunklocal
> include => iaxtel700
> include => trunktollfree
> include => iaxprovider
>
> [macro-stdexten]
> exten => s,1,Dial(${ARG2},20)
> exten => s,2,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(${ARG1},u)
> exten => s-NOANSWER,2,Goto(default,s,1)
> exten => s-BUSY,1,Voicemail(${ARG1},b)
> exten => s-BUSY,2,Goto(default,s,1)
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
>
> [macro-stdPrivacyexten]
> exten => s,1,Dial(${ARG2},20|p)
> exten => s,2,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
> exten => s-DONTCALL,1,Goto(${ARG3},s,1)
> exten => s-TORTURE,1,Goto(${ARG4},s,1)
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
>
> [macro-page]
> exten => s,1,ChanIsAvail(${ARG1}|js)
> exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
> exten => s,n(autoanswer),Set(_ALERT_INFO="RA")
> exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)
> exten => s,n,NoOp()
> exten => s,n,Dial(${ARG1}||)
> exten => s,n(fail),Hangup
>
> [demo]
> exten => s,1,Wait(1)
> exten => s,n,Answer
> exten => s,n,Set(TIMEOUT(digit)=5)
> exten => s,n,Set(TIMEOUT(response)=10)
> exten => s,n(restart),BackGround(demo-congrats)
> exten => s,n(instruct),BackGround(demo-instruct)
> exten => s,n,WaitExten
> exten => 2,1,BackGround(demo-moreinfo)
> exten => 2,n,Goto(s,instruct)
> exten => 3,1,Set(LANGUAGE()=fr)
> exten => 3,n,Goto(s,restart)
> exten => 1000,1,Goto(default,s,1)
> exten => 1234,1,Playback(transfer,skip)
> exten => 1234,n,Macro(stdexten,1234,${CONSOLE})
> exten => 1235,1,Voicemail(u1234)
> exten => 1236,1,Dial(Console/dsp)
> exten => 1236,n,Voicemail(u1234)
> exten => #,1,Playback(demo-thanks)
> exten => #,n,Hangup
> exten => t,1,Goto(#,1)
> exten => i,1,Playback(invalid)
> exten => 500,1,Playback(demo-abouttotry)
> exten => 500,n,Dial(IAX2/gu...@mi.../s@default)
> exten => 500,n,Playback(demo-nogo)
> exten => 500,n,Goto(s,6)
> exten => 600,1,Playback(demo-echotest)
> exten => 600,n,Echo
> exten => 600,n,Playback(demo-echodone)
> exten => 600,n,Goto(s,6)
> exten => 76245,1,Macro(page,SIP/Grandstream1)
> exten => _7XXX,1,Macro(page,SIP/${EXTEN})
> exten => 7999,1,Set(TIMEOUT(absolute)=60)
> exten =>
> 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/ 
> 1234@page/n|d)
> exten => 8500,1,VoicemailMain
> exten => 8500,n,Goto(s,6)
>
> [page]
> exten => _X.,1,Macro(page,SIP/${EXTEN})
>
> [voicemenu-custom-1]
> include = default
> comment = Welcome
> alias_exten = 7000
> exten = s,1,Answer
> exten = s,2,Wait(1)
> exten = s,3,Background(thank-you-for-calling)
> exten = s,4,Background(if-u-know-ext-dial)
> exten = s,5,Background(otherwise)
> exten = s,6,Background(to-reach-operator)
> exten = s,7,Background(pls-hold-while-try)
> exten = s,8,WaitExten(6)
> exten = 0,1,Goto(default|6010|1)
>
> [numberplan-custom-1]
> plancomment = DialPlan1
> include = default
> include = parkedcalls
> exten = _9011XXXXXXX!,1,Macro(trunkdial,${}/${EXTEN:1})
> comment = _9011XXXXXXX!,1,International,standard
> exten = _911!,1,Macro(trunkdial,${}/${EXTEN:0})
> comment = _911!,1,911,standard
> exten =
> _9256XXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:4},${trunk_1_cid})
> comment = _9256XXXXXXX!,1,Local,standard
> exten =
> _91XXXXXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1},${trunk_1_cid})
> comment = _91XXXXXXXXXX!,1,Longdistance,standard
> exten =
> _91700XXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1},${trunk_1_cid})
> comment = _91700XXXXXXX!,1,IAXTEL,standard
> exten = _8!,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid})
> comment = _8!,1,Oi,standard
> exten = _0XXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1},$ 
> {trunk_1_cid})
> comment = _0XXXXXXX!,1,Local,standard
> exten =
> _0800XXXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid})
> comment = _0800XXXXXXXX!,1,0800,standard
>
> [macro-trunkdial]
> exten = s,1,set(CALLERID(all)=${IF($["${LEN(${CALLERID(num)})}" >
> "6"]?${CALLERID(all)}:${ARG2})})
> exten = s,n,Dial(${ARG1})
> exten = s,n,Goto(s-${DIALSTATUS},1)
> exten = s-NOANSWER,1,Hangup
> exten = s-BUSY,1,Hangup
> exten = _s-.,1,NoOp
> exten = s-BUSY,1,Hangup
> exten = _s-.,1,NoOp
>
> [asterisk_guitools]
> exten = executecommand,1,System(${command})
> exten = executecommand,n,Hangup()
> exten = record_vmenu,1,Answer
> exten = record_vmenu,n,Playback(vm-intro)
> exten = record_vmenu,n,Record(${var1})
> exten = record_vmenu,n,Playback(vm-saved)
> exten = record_vmenu,n,Playback(vm-goodbye)
> exten = record_vmenu,n,Hangup
> exten = play_file,1,Answer
> exten = play_file,n,Playback(${var1})
> exten = play_file,n,Hangup
>
> [voicemenu-custom-2]
> comment = ura cobranca
> alias_exten = 9000
> exten = s,1,Answer
> exten = s,2,Background(you-wish-to-join)
>
> [default]
> exten = 6050,1,VoiceMailMain
> exten = 7000,1,Goto(voicemenu-custom-1|s|1)
> exten = 8000,1,MeetMe(${EXTEN}|MsI)
> exten = 6000,1,Queue(${EXTEN})
> exten = 9000,1,Goto(voicemenu-custom-2|s|1)
> exten = 6001,1,Queue(${EXTEN})
> exten = 1300,1,Agi(agi://localhost/hello.agi)
> exten = 6100,1,Goto(ringroups-custom-1|s|1)
> exten = 3030,1,Answer
> exten = 3030,2,Playback(vm-intro)
> exten = 3030,3,Wait(3)
> exten = 3030,4,Hangup
>
> [ringroups-custom-1]
> gui_ring_groupname = discador
> exten = s,1,NoOp(RINGGROUP)
> exten = s,n,Dial(SIP/6012&SIP/6016&SIP/6013&SIP/6014&SIP/6015,20)
> exten = s,n,Hangup
>
> [DID_trunk_1]
> include = default
> include = default
> include = default
> exten = _X.,1,Goto(default|6010|1)
> exten = s,1,ExecIf($[ "$ 
> {CALLERID(num)}"="" ],SetCallerPres,unavailable)
> exten = s,2,ExecIf($[ "$ 
> {CALLERID(num)}"="" ],Set,CALLERID(all)=unknown
> <0000000>)
> exten = s,3,Goto(default|6010|1)
>
> [DID_trunk_1]
>
> [DID_trunk_1]
>
> [DID_trunk_2]
> include = default
> ----------------------------------------------------------------------------
> -------------------------------
>
>
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