Re: [asterisk-users] pbx.c: We were unable to say the number
Hi Mohammad, which application do you use to say the number [n]? you can use *SayAlpha()*application to say the number you want. Please read link: *http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SayAlpha* for more infomation. Phuonghd On Sat, Mar 26, 2011 at 11:15 AM, Mohammad Khan beepl...@gmail.com wrote: Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check a number online or offline
Hi Dhaval,
Can you say how to fire action on AMI in this case and recieve response on
AMI. I also tried to do with HangupAction and RedirectAction action (using
asterisk-java library) in application java (AMI) to hang up or redirect a
channel that is online at the extension on asterisk but not successfully.
This is my code:
package Test;
import java.io.IOException;
import org.asteriskjava.manager.AuthenticationFailedException;
import org.asteriskjava.manager.ManagerConnection;
import org.asteriskjava.manager.ManagerConnectionFactory;
import org.asteriskjava.manager.TimeoutException;
import org.asteriskjava.manager.action.HangupAction;
import org.asteriskjava.manager.action.OriginateAction;
import org.asteriskjava.manager.action.RedirectAction;
import org.asteriskjava.manager.response.ManagerResponse;
public class TestOriginate {
/**
* @param args
*/
private ManagerConnection managerConnection;
public TestOriginate() throws IOException {
ManagerConnectionFactory factory = new ManagerConnectionFactory(
192.168.0.178, manager, pa55w0rd);
this.managerConnection = factory.createManagerConnection();
}
public void run() {
RedirectAction redirectAction;
ManagerResponse originateResponse;
String state = ;
String receiver = 0976468586;
redirectAction = new RedirectAction();
redirectAction.setContext(from-smg);
redirectAction.setExten(9220);
redirectAction.setPriority(new Integer(1));
redirectAction.setChannel(SIP/+ receiver);
try {
System.out.println(Starting login 192.168.0.178);
managerConnection.login();
System.out.println(After login 192.168.0.178);
} catch (IllegalStateException e) {
} catch (TimeoutException e) {
} catch (IOException e) {
} catch (AuthenticationFailedException e) {
}
try {
originateResponse = managerConnection.sendAction(redirectAction,
3);
state = originateResponse.getResponse();
System.out.println(State value is : + state);
} catch (IllegalArgumentException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (IllegalStateException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (TimeoutException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
managerConnection.logoff();
}
public static void main(String[] args) throws IOException {
// TODO Auto-generated method stub
TestOriginate test = new TestOriginate();
test.run();
}
}
*While i run above code, the result printed on console likes following:*
Starting login 192.168.0.178
Jan 11, 2011 3:26:01 PM
org.asteriskjava.manager.internal.ManagerConnectionImpl connect
INFO: Connecting to 192.168.0.178:5038
Jan 11, 2011 3:26:02 PM
org.asteriskjava.manager.internal.ManagerConnectionImpl
setProtocolIdentifier
INFO: Connected via Asterisk Call Manager/1.1
Jan 11, 2011 3:26:02 PM
org.asteriskjava.manager.internal.ManagerConnectionImpl
setProtocolIdentifier
WARNING: Unsupported protocol version 'Asterisk Call Manager/1.1'. Use at
your own risk.
Jan 11, 2011 3:26:02 PM
org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
INFO: Successfully logged in
Jan 11, 2011 3:26:04 PM
org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
INFO: Determined Asterisk version: Asterisk 1.0
After login 192.168.0.178
State value is :Error
Jan 11, 2011 3:26:04 PM
org.asteriskjava.manager.internal.ManagerConnectionImpl disconnect
INFO: Closing socket.
Jan 11, 2011 3:26:04 PM org.asteriskjava.manager.internal.ManagerReaderImpl
run
INFO: Terminating reader thread: socket closed
I hope you can spend your time to read what i have written above and help me
solve this problem.
Can you contact with me by my yahoo nick : ducphuongbk200...@yahoo.com
Thanks and best regards.
Phuong
On Mon, Jan 10, 2011 at 10:48 PM, DHAVAL INDRODIYA dhaval.it01...@gmail.com
wrote:
HI Phuong,
JIM is right way but if you want to use extension state then there is a
simple way of achiving through
AMI, you need to fire this action on AMI and response have your answer ,
Please read about Action ExtensionState.
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+ExtensionState
If you are looking for extension state just pass extension and you will
receive perfect response of that extension then you cans code as you want.
regards
Dhaval
On Tue, Jan 11, 2011 at 9:56 AM, Phuong Hoang ducphuongbk200...@gmail.com
wrote:
Hi Jim,
Really, I have`nt understood what you said yet. I am building a system on
asterisk, and want to check a number online
Re: [asterisk-users] How to check a number online or offline
Hi Dhaval,
I fired originate action on AMI and everything is ok but redirect action not
ok.
here the channel i set is available and context also exists on file
extension.conf .
I will send manager.conf,extensions.conf and sip.conf (in Extension.rar) to
you.
I registered a sip phone with account *0976468586 *and context *TestAMQ*
on asterisk =ok
When i call with extension of *TestAMQ*(*this case extension is 999* ) by
sip phone account* 0976468586* and simultaneously run above java program
(AMI) that i sent to you to redirect the number *0976468586* to context *
from-smg* then received the error.
Do you use yahoo or skype?if you do, can you let me know *your ID yahoo or
skype* to say something easier. My Yahoo ID is : *
ducphuongbk200...@yahoo.com*
Thanks and best regard
Phuong
On Tue, Jan 11, 2011 at 2:05 AM, Phuong Hoang
ducphuongbk200...@gmail.comwrote:
On Tue, Jan 11, 2011 at 1:18 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi Phuong,
i see your code is looking nice and there is no problem in implementation
, if you have any problem
then first send me manager.conf file then try to connect through manager
using telnet and then fire same action on this in that you can get proper
error codes .
one more thing the channel you set is this channel is available to
redirected???
regards
Dhavak
On Tue, Jan 11, 2011 at 2:05 PM, Phuong Hoang
ducphuongbk200...@gmail.com wrote:
Hi Dhaval,
Can you say how to fire action on AMI in this case and recieve response
on AMI. I also tried to do with HangupAction and RedirectAction action
(using asterisk-java library) in application java (AMI) to hang up or
redirect a channel that is online at the extension on asterisk but not
successfully. This is my code:
package Test;
import java.io.IOException;
import org.asteriskjava.manager.AuthenticationFailedException;
import org.asteriskjava.manager.ManagerConnection;
import org.asteriskjava.manager.ManagerConnectionFactory;
import org.asteriskjava.manager.TimeoutException;
import org.asteriskjava.manager.action.HangupAction;
import org.asteriskjava.manager.action.OriginateAction;
import org.asteriskjava.manager.action.RedirectAction;
import org.asteriskjava.manager.response.ManagerResponse;
public class TestOriginate {
/**
* @param args
*/
private ManagerConnection managerConnection;
public TestOriginate() throws IOException {
ManagerConnectionFactory factory = new ManagerConnectionFactory(
192.168.0.178, manager, pa55w0rd);
this.managerConnection = factory.createManagerConnection();
}
public void run() {
RedirectAction redirectAction;
ManagerResponse originateResponse;
String state = ;
String receiver = 0976468586;
redirectAction = new RedirectAction();
redirectAction.setContext(from-smg);
redirectAction.setExten(9220);
redirectAction.setPriority(new Integer(1));
redirectAction.setChannel(SIP/+ receiver);
try {
System.out.println(Starting login 192.168.0.178);
managerConnection.login();
System.out.println(After login 192.168.0.178);
} catch (IllegalStateException e) {
} catch (TimeoutException e) {
} catch (IOException e) {
} catch (AuthenticationFailedException e) {
}
try {
originateResponse =
managerConnection.sendAction(redirectAction,
3);
state = originateResponse.getResponse();
System.out.println(State value is : + state);
} catch (IllegalArgumentException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (IllegalStateException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (TimeoutException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
managerConnection.logoff();
}
public static void main(String[] args) throws IOException {
// TODO Auto-generated method stub
TestOriginate test = new TestOriginate();
test.run();
}
}
*While i run above code, the result printed on console likes following:
*
Starting login 192.168.0.178
Jan 11, 2011 3:26:01 PM
org.asteriskjava.manager.internal.ManagerConnectionImpl connect
INFO: Connecting to 192.168.0.178:5038
Jan 11, 2011 3:26:02 PM
org.asteriskjava.manager.internal.ManagerConnectionImpl
setProtocolIdentifier
INFO: Connected via Asterisk Call Manager/1.1
Jan 11, 2011 3:26:02 PM
org.asteriskjava.manager.internal.ManagerConnectionImpl
setProtocolIdentifier
WARNING: Unsupported protocol version 'Asterisk Call Manager/1.1'. Use
at your own risk.
Jan 11, 2011 3
Re: [asterisk-users] How to check a number online or offline
Hi Dhaval,
I have`nt known you recieved my mail yet?if you did and you can answer my
question then please rely to you, i am looking forward to listening your
reply.
Have a good nice.
Thanks and best regards!
Phuong.
On Tue, Jan 11, 2011 at 2:11 AM, Phuong Hoang
ducphuongbk200...@gmail.comwrote:
Hi Dhaval,
I fired originate action on AMI and everything is ok but redirect action
not ok.
here the channel i set is available and context also exists on file
extension.conf .
I will send manager.conf,extensions.conf and sip.conf (in Extension.rar) to
you.
I registered a sip phone with account *0976468586 *and context *TestAMQ*
on asterisk =ok
When i call with extension of *TestAMQ*(*this case extension is 999* )
by sip phone account* 0976468586* and simultaneously run above java
program (AMI) that i sent to you to redirect the number *0976468586* to
context *from-smg* then received the error.
Do you use yahoo or skype?if you do, can you let me know *your ID yahoo
or skype* to say something easier. My Yahoo ID is : *
ducphuongbk200...@yahoo.com*
Thanks and best regard
Phuong
On Tue, Jan 11, 2011 at 2:05 AM, Phuong Hoang ducphuongbk200...@gmail.com
wrote:
On Tue, Jan 11, 2011 at 1:18 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi Phuong,
i see your code is looking nice and there is no problem in
implementation , if you have any problem
then first send me manager.conf file then try to connect through manager
using telnet and then fire same action on this in that you can get proper
error codes .
one more thing the channel you set is this channel is available to
redirected???
regards
Dhavak
On Tue, Jan 11, 2011 at 2:05 PM, Phuong Hoang
ducphuongbk200...@gmail.com wrote:
Hi Dhaval,
Can you say how to fire action on AMI in this case and recieve response
on AMI. I also tried to do with HangupAction and RedirectAction action
(using asterisk-java library) in application java (AMI) to hang up or
redirect a channel that is online at the extension on asterisk but not
successfully. This is my code:
package Test;
import java.io.IOException;
import org.asteriskjava.manager.AuthenticationFailedException;
import org.asteriskjava.manager.ManagerConnection;
import org.asteriskjava.manager.ManagerConnectionFactory;
import org.asteriskjava.manager.TimeoutException;
import org.asteriskjava.manager.action.HangupAction;
import org.asteriskjava.manager.action.OriginateAction;
import org.asteriskjava.manager.action.RedirectAction;
import org.asteriskjava.manager.response.ManagerResponse;
public class TestOriginate {
/**
* @param args
*/
private ManagerConnection managerConnection;
public TestOriginate() throws IOException {
ManagerConnectionFactory factory = new
ManagerConnectionFactory(
192.168.0.178, manager, pa55w0rd);
this.managerConnection = factory.createManagerConnection();
}
public void run() {
RedirectAction redirectAction;
ManagerResponse originateResponse;
String state = ;
String receiver = 0976468586;
redirectAction = new RedirectAction();
redirectAction.setContext(from-smg);
redirectAction.setExten(9220);
redirectAction.setPriority(new Integer(1));
redirectAction.setChannel(SIP/+ receiver);
try {
System.out.println(Starting login 192.168.0.178);
managerConnection.login();
System.out.println(After login 192.168.0.178);
} catch (IllegalStateException e) {
} catch (TimeoutException e) {
} catch (IOException e) {
} catch (AuthenticationFailedException e) {
}
try {
originateResponse =
managerConnection.sendAction(redirectAction,
3);
state = originateResponse.getResponse();
System.out.println(State value is : + state);
} catch (IllegalArgumentException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (IllegalStateException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (TimeoutException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
managerConnection.logoff();
}
public static void main(String[] args) throws IOException {
// TODO Auto-generated method stub
TestOriginate test = new TestOriginate();
test.run();
}
}
*While i run above code, the result printed on console likes
following:*
Starting login 192.168.0.178
Jan 11, 2011 3:26:01 PM
org.asteriskjava.manager.internal.ManagerConnectionImpl connect
INFO: Connecting to 192.168.0.178:5038
Jan 11, 2011 3:26:02 PM
Re: [asterisk-users] How to reject an incoming call using AMI ?
Hi Olivier, I don`t really understand what you said. Actually, the issue that i face on is i don`t know how to redirect a number online on the context (example testA context) to other context (example testB context). Can you help me to solve this issue. Thanks and best regard. Phuong On Tue, Jan 11, 2011 at 12:04 AM, Olivier oza_4...@yahoo.fr wrote: 2011/1/11 Phuong Hoang ducphuongbk200...@gmail.com Hi Rodrigo, Can you say clearlier about using command Hangup in the AMI to reject or hang up a incoming call?I also have the same issue. Thanks and looks forward to listening your reply soon! Best regards, Phuong On Mon, Jan 10, 2011 at 2:13 PM, Rodrigo Lang rodrigoferreiral...@gmail.com wrote: Hi. You see the comando Hangup in the AMI? Best regards, Rodrigo Lang. 2011/1/10 Olivier oza_4...@yahoo.fr Hi, For a call center, I'm studying how I can offer agents the ability to reject an incoming call using a custom application. As you can guess, in this case, rejecting a call means let another agent answer this call (it doesn't mean end this call). The only way I could imagine for this to happen, would be to redirect the caller to a conference room, then hangup the agent call leg and then redirect the caller back to the appropriate queue, hoping the caller wouldn't be once again forwarded to the busy agent. Which way to implement this would you suggest or recommend ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My understanding is: if using AMI or console, you hangup one call leg, both legs will be hanged up. In this case, I'm looking for a way to reproduce IP phones Reject feature. With this feature, a ringing SIP phone would reply to an incoming call with Busy signal which will be treated as such by Asterisk. Hopefully, this Busy reply would let the incoming go to the next available agent. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reject an incoming call using AMI ?
Sorry,I forget redirecting from this context to that context is done by an java application(AMI). On Tue, Jan 11, 2011 at 7:55 AM, Phuong Hoang ducphuongbk200...@gmail.comwrote: Hi Olivier, I don`t really understand what you said. Actually, the issue that i face on is i don`t know how to redirect a number online on the context (example testA context) to other context (example testB context). Can you help me to solve this issue. Thanks and best regard. Phuong On Tue, Jan 11, 2011 at 12:04 AM, Olivier oza_4...@yahoo.fr wrote: 2011/1/11 Phuong Hoang ducphuongbk200...@gmail.com Hi Rodrigo, Can you say clearlier about using command Hangup in the AMI to reject or hang up a incoming call?I also have the same issue. Thanks and looks forward to listening your reply soon! Best regards, Phuong On Mon, Jan 10, 2011 at 2:13 PM, Rodrigo Lang rodrigoferreiral...@gmail.com wrote: Hi. You see the comando Hangup in the AMI? Best regards, Rodrigo Lang. 2011/1/10 Olivier oza_4...@yahoo.fr Hi, For a call center, I'm studying how I can offer agents the ability to reject an incoming call using a custom application. As you can guess, in this case, rejecting a call means let another agent answer this call (it doesn't mean end this call). The only way I could imagine for this to happen, would be to redirect the caller to a conference room, then hangup the agent call leg and then redirect the caller back to the appropriate queue, hoping the caller wouldn't be once again forwarded to the busy agent. Which way to implement this would you suggest or recommend ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My understanding is: if using AMI or console, you hangup one call leg, both legs will be hanged up. In this case, I'm looking for a way to reproduce IP phones Reject feature. With this feature, a ringing SIP phone would reply to an incoming call with Busy signal which will be treated as such by Asterisk. Hopefully, this Busy reply would let the incoming go to the next available agent. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check a number online or offline
Thanks enkillar, but this is`nt thing that i need. I want to check number online, offline or unreachable on asterisk using AMI(Asterisk Manager Interface) by java but i have`nt found a solution yet. I hope you can help me do this. Thanks in advance ! On Mon, Jan 10, 2011 at 1:54 AM, enkillar 87 enkil...@live.com wrote: Hi Phuong, From Asterisk CLI, run command: sip show peers Regards, Huy Nguyen -- Date: Sun, 9 Jan 2011 20:01:57 -0800 From: ducphuongbk200...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to check a number online or offline Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check a number online or offline
Thanks Steve Howes, I found the link you have just sent to me but it do`nt help me to resolve this. Can you say clearlier for me? Thanks so much! On Mon, Jan 10, 2011 at 2:21 AM, Steve Howes steve-li...@geekinter.netwrote: On 10 Jan 2011, at 10:17, Phuong Hoang wrote: Thanks enkillar, but this is`nt thing that i need. I want to check number online, offline or unreachable on asterisk using AMI(Asterisk Manager Interface) by java but i have`nt found a solution yet. I hope you can help me do this. Thanks in advance ! http://www.voip-info.org/wiki/view/Asterisk+manager+API There are a number of commands there that would help if you'd bothered to look.. Can retrieve sip peers with one, or a generic 'command' command would do it too in most cases.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check a number online or offline
Thanks Dhaval, My purpose is that i want to use java application (using Asterisk Manager Interface) to check a number online, offline or unreachable. Your suggest uses function DEVICE_STATE but this is written in dialplan not application java. Do you know other way to do this for me?thanks and looks forward to listening your reply. Regards! Phuong On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: Hello , You can use Dialplan function DEVICE_STATE, which will gives you perfect status of DEVICE. regards Dhaval On Mon, Jan 10, 2011 at 4:11 PM, Steve Howes steve-li...@geekinter.netwrote: On 10 Jan 2011, at 10:37, Phuong Hoang wrote: I found the link you have just sent to me but it do`nt help me to resolve this. Can you say clearlier for me? Not really. It's a list of manager commands. There is 'SIPshowpeer' which will work for sip stuff. Try the command 'Command' action and you can send any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work in some cases.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reject an incoming call using AMI ?
Hi Rodrigo, Can you say clearlier about using command Hangup in the AMI to reject or hang up a incoming call?I also have the same issue. Thanks and looks forward to listening your reply soon! Best regards, Phuong On Mon, Jan 10, 2011 at 2:13 PM, Rodrigo Lang rodrigoferreiral...@gmail.com wrote: Hi. You see the comando Hangup in the AMI? Best regards, Rodrigo Lang. 2011/1/10 Olivier oza_4...@yahoo.fr Hi, For a call center, I'm studying how I can offer agents the ability to reject an incoming call using a custom application. As you can guess, in this case, rejecting a call means let another agent answer this call (it doesn't mean end this call). The only way I could imagine for this to happen, would be to redirect the caller to a conference room, then hangup the agent call leg and then redirect the caller back to the appropriate queue, hoping the caller wouldn't be once again forwarded to the busy agent. Which way to implement this would you suggest or recommend ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check a number online or offline
Thanks Jim, Can you say about your idea clearlier? I want to use AMI in an application java to check a number online, offline or unreachable and result is returned to the appliction java. If the number is online now, i will use AMI to hangup it, else i do nothing. Best regards, Phuong. On Mon, Jan 10, 2011 at 8:50 AM, Jim Dickenson dicken...@cfmc.com wrote: You can always place a call to an extension that sends a user event from AMI. If there are no native AMI commands that can return what you want originate a call to a local extension that returns a user event. -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 10, 2011, at 7:48 AM, Phuong Hoang wrote: Thanks Dhaval, My purpose is that i want to use java application (using Asterisk Manager Interface) to check a number online, offline or unreachable. Your suggest uses function DEVICE_STATE but this is written in dialplan not application java. Do you know other way to do this for me?thanks and looks forward to listening your reply. Regards! Phuong On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hello , You can use Dialplan function DEVICE_STATE, which will gives you perfect status of DEVICE. regards Dhaval On Mon, Jan 10, 2011 at 4:11 PM, Steve Howes steve-li...@geekinter.netwrote: On 10 Jan 2011, at 10:37, Phuong Hoang wrote: I found the link you have just sent to me but it do`nt help me to resolve this. Can you say clearlier for me? Not really. It's a list of manager commands. There is 'SIPshowpeer' which will work for sip stuff. Try the command 'Command' action and you can send any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work in some cases.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check a number online or offline
Hi Jim,
Really, I have`nt understood what you said yet. I am building a system on
asterisk, and want to check a number online, offline or unreachable. If
number is online on the extension then i want to redirect other extension.
Redirecting is done by application java using AMI. can you help me do it?
Thanks and best regards!
Phuong
On Mon, Jan 10, 2011 at 7:09 PM, Jim Dickenson dicken...@cfmc.com wrote:
If you do an AMI packet like this:
Action: Originate
Channel: Local/get_i...@some_context
Exten: do_noop
Context: some_context
Priority: 1
ActionID: GetInfo
Async: true
and then have a couple extensions that do what you want. Here is what I do
in my case:
exten = get_info,1,Answer()
exten = get_info,n,UserEvent(GetInfo,Version:ABE
DateTime:${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)} CfMC:83351)
exten = get_info,n,Hangup()
exten = do_noop,1,Answer()
exten = do_noop,n,Wait(1)
exten = do_noop,n,Hangup()
You would then do what you need to do in your extensions.
--
Jim Dickenson
mailto:dicken...@cfmc.com dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 10, 2011, at 6:16 PM, Phuong Hoang wrote:
Thanks Jim,
Can you say about your idea clearlier? I want to use AMI in an application
java to check a number online, offline or unreachable and result is returned
to the appliction java. If the number is online now, i will use AMI to
hangup it, else i do nothing.
Best regards,
Phuong.
On Mon, Jan 10, 2011 at 8:50 AM, Jim Dickenson dicken...@cfmc.com wrote:
You can always place a call to an extension that sends a user event from
AMI. If there are no native AMI commands that can return what you want
originate a call to a local extension that returns a user event.
--
Jim Dickenson
mailto:dicken...@cfmc.com dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 10, 2011, at 7:48 AM, Phuong Hoang wrote:
Thanks Dhaval,
My purpose is that i want to use java application (using Asterisk Manager
Interface) to check a number online, offline or unreachable. Your suggest
uses function DEVICE_STATE but this is written in dialplan not application
java. Do you know other way to do this for me?thanks and looks forward to
listening your reply.
Regards!
Phuong
On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hello ,
You can use Dialplan function DEVICE_STATE, which will gives you perfect
status of DEVICE.
regards
Dhaval
On Mon, Jan 10, 2011 at 4:11 PM, Steve Howes
steve-li...@geekinter.netwrote:
On 10 Jan 2011, at 10:37, Phuong Hoang wrote:
I found the link you have just sent to me but it do`nt help me to
resolve this. Can you say clearlier for me?
Not really. It's a list of manager commands. There is 'SIPshowpeer'
which will work for sip stuff. Try the command 'Command' action and you can
send any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might
work in some cases..
S
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[asterisk-users] How to check a number online or offline
Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit
-- Forwarded message -- From: Phuong Hoang ducphuongbk200...@gmail.com Date: Thu, Nov 18, 2010 at 9:16 AM Subject: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit To: asterisk-users@lists.digium.com Hi all, I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Thanks and best regards. Phuong Hoang. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to register SIP phone on Asterisk 1.6.2.14 on Centos 5.5 64bit
Hi all, I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Thanks and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit
Hi all, I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Thanks and best regards. Phuong Hoang. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to construct a call center on asterisk
Hi all Now, i want to construct a call center on asterisk but i don`t know how to do. Can anyone help me? Thanks and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to construct a call center on asterisk
Thanks Cary, I constructed IVR (*Interactive Voice Response*) system on asterisk and now i want to construct a call center to support IVR system.The call center that i want to construct has 7-8 employees so that i don`t want to buy it. I found on internet and guided using Open SIP Server with asterisk to construct a call center but i don`t know how to do now. Do you know using Open SIP Server with asterisk to do?please help me! Thanks and best regards. Phuong Hoang. On Tue, Nov 16, 2010 at 10:32 AM, Cary Fitch ca...@usawide.net wrote: That is a broad question. If it is a 10-20 person call center, you may do it. If it is a high density center with lots of lines, requiring fail over capability, etc. so that hundreds of employees are not sitting around during down time, I would suggest “designing it” by buying it from a manufacturer that has done it before. I know of a company whose name starts with “X” . There may be others. Cary Fitch -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Phuong Hoang *Sent:* Monday, November 15, 2010 8:01 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] How to construct a call center on asterisk Hi all Now, i want to construct a call center on asterisk but i don`t know how to do. Can anyone help me? Thanks and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Store CDR (call detail record) to Oracle database
Thanks Danny. I searched on internet and found a link : http://forums.digium.com/viewtopic.php?f=1t=11135start=0 This link guided to use ODBC to store cdr into Oracle database but not successful . Can you help me ? Thanks and best regards. On Tue, Nov 9, 2010 at 9:27 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Phuong Hoang *Sent:* Monday, November 08, 2010 8:31 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Store CDR (call detail record) to Oracle database Hi all, Now i want to store cdr (call detail record) to Oracle database but i don't know how to do .Can anyone help me ? Thanks and best regards. ODBC is going to be your best bet. At some point you might be able to use the MYSQL stuff to talk from Asterisk to Oracle, but for now I’m pretty sure it is only accessible through ODBC. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Store CDR (call detail record) to Oracledatabase
Thanks Danny. Can you talk clearer about using ODBC to store cdr to oracle db because i don`t want to synchronize oracle db with mysql db? Thanks and best regards. On Wed, Nov 10, 2010 at 9:15 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Phuong Hoang *Sent:* Wednesday, November 10, 2010 3:07 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Store CDR (call detail record) to Oracledatabase Thanks Danny. I searched on internet and found a link : http://forums.digium.com/viewtopic.php?f=1t=11135start=0 This link guided to use ODBC to store cdr into Oracle database but not successful . Can you help me ? Thanks and best regards. On Tue, Nov 9, 2010 at 9:27 PM, Danny Nicholas da...@debsinc.com wrote: -- ODBC is going to be your best bet. At some point you might be able to use the MYSQL stuff to talk from Asterisk to Oracle, but for now I’m pretty sure it is only accessible through ODBC. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Phuong Hoang *Sent:* Monday, November 08, 2010 8:31 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Store CDR (call detail record) to Oracle database Hi all, Now i want to store cdr (call detail record) to Oracle database but i don't know how to do .Can anyone help me ? Thanks and best regards. Just make your Oracle tables using the MYSQL guidelines and you should be good to go. If it proves to be too cumbersome using odbc, you could always make your “live” cdr in flat file or MYSQL and do a quick-and-dirty daemon to synchronize that to the Oracle DB. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Store CDR (call detail record) toOracledatabase
Thanks for your reply.now i decide to read from flat file /var/log/asterisk/cdr-csv/Master.csv and then insert into oracle database. it has been to meet you. On Wed, Nov 10, 2010 at 11:46 PM, Danny Nicholas da...@debsinc.com wrote: I’d love to, but I have bigger trees to chop in Asterisk that messing with DB CDR’s, so I just use the flat file /var/log/asterisk/cdr-csv/Master.csv. I’ll defer this answer to a poster who has actually climbed that mountain. Apologies for terse answer and top-post. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Phuong Hoang *Sent:* Wednesday, November 10, 2010 10:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Store CDR (call detail record) toOracledatabase Thanks Danny. Can you talk clearer about using ODBC to store cdr to oracle db because i don`t want to synchronize oracle db with mysql db? Thanks and best regards. On Wed, Nov 10, 2010 at 9:15 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Phuong Hoang *Sent:* Wednesday, November 10, 2010 3:07 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Store CDR (call detail record) to Oracledatabase Thanks Danny. I searched on internet and found a link : http://forums.digium.com/viewtopic.php?f=1t=11135start=0 This link guided to use ODBC to store cdr into Oracle database but not successful . Can you help me ? Thanks and best regards. On Tue, Nov 9, 2010 at 9:27 PM, Danny Nicholas da...@debsinc.com wrote: -- ODBC is going to be your best bet. At some point you might be able to use the MYSQL stuff to talk from Asterisk to Oracle, but for now I’m pretty sure it is only accessible through ODBC. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Phuong Hoang *Sent:* Monday, November 08, 2010 8:31 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Store CDR (call detail record) to Oracle database Hi all, Now i want to store cdr (call detail record) to Oracle database but i don't know how to do .Can anyone help me ? Thanks and best regards. Just make your Oracle tables using the MYSQL guidelines and you should be good to go. If it proves to be too cumbersome using odbc, you could always make your “live” cdr in flat file or MYSQL and do a quick-and-dirty daemon to synchronize that to the Oracle DB. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Store CDR (call detail record) to Oracle database
Hi all, Now i want to store cdr (call detail record) to Oracle database but i don't know how to do .Can anyone help me ? Thanks and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
