Re: [asterisk-users] How to reject an incoming call using AMI ?
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
[asterisk-users] asterisk fax problem
Hello, I have asterisk 1.6.2.9-2 I tried to install fax utility as it is shown on pdf documents on asterisk site. I downloaded Opteron compiled res_fax and res_fax_digium files and copied to /usr/lib/asterisk/modules/ where default modules directory is. I created a free fax license and created license file on asterisk server too. WHen i run asterisk it crashed. I noticed that if res_fax.so file is in directory (even with _digium file or not) i got error already have an application 'RECEIVE Fax' and 'SEND Fax' If i remove res_fax.so file from directory asterisk loads but i got the following error msgs on debug [Jan 11 09:07:03] WARNING[11898]: loader.c:435 load_dynamic_module: Error loading module 'res_fax_digium': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_register [Jan 11 09:07:03] WARNING[11898]: loader.c:801 load_resource: Module 'res_fax_digium' could not be loaded. Cam sonebody help me about what is going on :) -- _ -- 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
Re: [asterisk-users] asterisk fax problem
On Tuesday, January 11, 2011 10:14:28 am Oguzhan Kayhan wrote: Hello, I have asterisk 1.6.2.9-2 I tried to install fax utility as it is shown on pdf documents on asterisk site. I downloaded Opteron compiled res_fax and res_fax_digium files and copied to /usr/lib/asterisk/modules/ where default modules directory is. I created a free fax license and created license file on asterisk server too. WHen i run asterisk it crashed. I noticed that if res_fax.so file is in directory (even with _digium file or not) i got error already have an application 'RECEIVE Fax' and 'SEND Fax' If i remove res_fax.so file from directory asterisk loads but i got the following error msgs on debug [Jan 11 09:07:03] WARNING[11898]: loader.c:435 load_dynamic_module: Error loading module 'res_fax_digium': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_register [Jan 11 09:07:03] WARNING[11898]: loader.c:801 load_resource: Module 'res_fax_digium' could not be loaded. Cam sonebody help me about what is going on :) -- _ -- 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 An update I didnt restart asterisk after copying res_fax and res_fax_digium files. Instead i just did module load res_fax I got following msg. [Jan 11 09:35:22] NOTICE[12269]: res_fax.c:2526 load_module: Generic FAX application module version 1.6.2.0_1.2.1, Copyright (C) 2008-2009 Digium, Inc. [Jan 11 09:35:22] NOTICE[12269]: res_fax.c:2529 load_module: This module is supplied under a commercial license granted by Digium, Inc. [Jan 11 09:35:22] NOTICE[12269]: res_fax.c:2530 load_module: Please see the full license text supplied by the accompanying [Jan 11 09:35:22] NOTICE[12269]: res_fax.c:2531 load_module: register utility, or ask for a copy from Digium. [Jan 11 09:35:22] NOTICE[12269]: res_fax.c:2290 set_config: Configuration file 'res_fax.conf' not found, using default options. [Jan 11 09:35:22] WARNING[12269]: pbx.c:5085 ast_register_application2: Already have an application 'SendFAX' [Jan 11 09:35:22] WARNING[12269]: res_fax.c:2548 load_module: failed to register 'SendFAX'. asterisk*CLI module load res_fax_digium Loaded res_fax_digium [Jan 11 09:35:28] NOTICE[12269]: res_fax_digium.c:2495 load_module: Digium FAX technology module version 1.6.2.0_1.2.1, Copyright (C) 2008-2009 Digium, Inc. [Jan 11 09:35:28] NOTICE[12269]: res_fax_digium.c:2498 load_module: This module is supplied under a commercial license granted by Digium, Inc. [Jan 11 09:35:28] NOTICE[12269]: res_fax_digium.c:2499 load_module: Please see the full license text supplied by the accompanying [Jan 11 09:35:28] NOTICE[12269]: res_fax_digium.c:2500 load_module: register utility, or ask for a copy from Digium. [Jan 11 09:35:28] NOTICE[12269]: res_fax_digium.c:2503 load_module: This product includes software developed by the OpenSSL Project [Jan 11 09:35:28] NOTICE[12269]: res_fax_digium.c:2504 load_module: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Jan 11 09:35:28] NOTICE[12269]: res_fax_digium.c:2505 load_module: Copyright (C) 1998-2008 The OpenSSL Project == Adding single Free FAX For Asterisk license But if i restart asterisk, it hangs again. I remove res_FAx.so and it can be started again. But now i can not load res_Fax_digium module. then i ran following command : -- _ -- 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 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.comwrote:
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
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
Re: [asterisk-users] asterisk fax problem
On 01/11/2011 06:48 AM, Steve Underwood wrote: On 01/11/2011 04:14 PM, Oguzhan Kayhan wrote: Hello, I have asterisk 1.6.2.9-2 I tried to install fax utility as it is shown on pdf documents on asterisk site. I downloaded Opteron compiled res_fax and res_fax_digium files and copied to /usr/lib/asterisk/modules/ where default modules directory is. I created a free fax license and created license file on asterisk server too. WHen i run asterisk it crashed. I noticed that if res_fax.so file is in directory (even with _digium file or not) i got error already have an application 'RECEIVE Fax' and 'SEND Fax' If i remove res_fax.so file from directory asterisk loads but i got the following error msgs on debug [Jan 11 09:07:03] WARNING[11898]: loader.c:435 load_dynamic_module: Error loading module 'res_fax_digium': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_register [Jan 11 09:07:03] WARNING[11898]: loader.c:801 load_resource: Module 'res_fax_digium' could not be loaded. Cam sonebody help me about what is going on :) Install spandsp as your FAX engine, and make your problems go away. He already has it, he has app_fax loaded by default (by whatever Asterisk packages he is using, since 1.6.2.9-2 is not an official Asterisk version number). Oguzhan: Your Asterisk installation already has SendFAX and ReceiveFAX applications (based on Steve's excellent spandsp library) installed, and you can use them right now. If you want to try Digium's Fax For Asterisk modules, you'll have to add noload = app_fax.so to your /etc/asterisk/modules.conf file, so that you don't have two modules trying to provide the same SendFAX/ReceiveFAX applications. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] OpenVPN + SIP configuration?
Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting through Ethernet in hotels or public wifi hotspots. By any chance, would someone have a working configuration so I can take a look? Thank you. -- _ -- 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] OpenVPN + SIP configuration?
On Tue, Jan 11, 2011 at 11:20 AM, Gilles codecompl...@free.fr wrote: Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting through Ethernet in hotels or public wifi hotspots. By any chance, would someone have a working configuration so I can take a look? Thank you. Lazy way would be to use http://www.zentyal.org/ and point and click your way there... * Number one issue with Microsoft Windows clients on OpenVPN is getting the routing right. ~~~ Andrew lathama Latham lath...@gmail.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
Re: [asterisk-users] OpenVPN + SIP configuration?
On Tue, Jan 11, 2011 at 9:29 AM, Andrew Latham lath...@gmail.com wrote: On Tue, Jan 11, 2011 at 11:20 AM, Gilles codecompl...@free.fr wrote: Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting through Ethernet in hotels or public wifi hotspots. By any chance, would someone have a working configuration so I can take a look? Thank you. Lazy way would be to use http://www.zentyal.org/ and point and click your way there... * Number one issue with Microsoft Windows clients on OpenVPN is getting the routing right. Verify that you have an end-to-end connection before trying to push any data through it. If you are running windows vista or windows 7 and start the connection with the OpenVPN GUI, you have to run it as administrator or it doesn't have the rights to add a route to the routing table. Don't be afraid of the certificate based method; it's really not hard! Using the shared secret will only allow a single point to point connection. That is, you have to use certificates if you want more than one client. -M -- _ -- 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] OpenVPN + SIP configuration?
Hi, I have OpenVPN and Asterisk working nicely. However, I do use certificates. Though, it shouldn't matter. Can you explain what doesn't work for you? Is the connection not established or is the Asterisk and it's client not communicating? -Bruce On Tue, Jan 11, 2011 at 9:20 AM, Gilles codecompl...@free.fr wrote: Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting through Ethernet in hotels or public wifi hotspots. By any chance, would someone have a working configuration so I can take a look? Thank you. -- _ -- 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] Call queues on load-balanced asterisks
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B. Christensen Sent: Tuesday, January 11, 2011 5:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call queues on load-balanced asterisks Hello Dhaval (and others), As far as I can tell, realtime queue will not solve my problem. I can statically define the same queue with the same members on two machines as well. I was planning to use realtime anyway. The issue is the actual queueing of the incoming calls. Let's say I define the queue IT-support with members Local/100 and Local/101 on both machines. The first call comes in and is distributed by Kamailio to Asterisk A, and answered by 100. The next call comes in to Asterisk B, and is answered by 101. At this point, both members are busy. Call 3 now comes in and is sent to Asterisk A, where it waits for a free member. Call 4 comes in and is also sent to Asterisk A, as is Call 5. Then call 6 is sent to Asterisk B. At this point 100 finishes his call and becomes free. Which call is delivered to 100? As far as I can tell, that's a 50/50 chance between call 3 and call 6. This is not correct behaviour! Call 6 should wait until calls 3, 4 and 5 (from the other server) have all been delivered. In the example above: When call 3 comes in, Asterisk A may even try to deliver it to 101, who gets call waiting indication. He will now have two simultaneous calls from the same queue! I have not found any way to share information about calls waiting in the queue, wait times, member states and so on between the two servers. Unless you guys know of a way, I think I'm going to have to ask the customer to change their design to master-slave (with failover) instead of load-balanced. With kind regards, Pan IMO your best solution to this is going to be using a database and AGI query to keep a quasi-real (delayed by a few ms/sec) picture of the queue activity. If you kept a database on both machines and ran an AGI with each incoming call to query queue usage on both machines or better yet, query the queue on the remote machine and spawn a short local call to keep that agent busy on the native machine, that would solve this issue. Let's say that a typical agent interaction occurs in 60 second chunks. Call 1 comes in to machine 1 and is answered by agent 100 as you said. Call 2 comes into machine 2 and is answered by 101. When Call 3 comes in, it sees 101 and 102 as busy on both machines. You can do this, but isn't this really a Kamailio issue? -- _ -- 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] Do I need a sip proxy?
Thanks a lot for the great input Pan. I think you are right on point with this one. I have STATIC PORT enabled in my outbound WAN. I am not sure if it was set for SIP or OpenVPN use but it is there for a reason. So, I try to mingle a bit with Siproxd package. I am a bit fuzzy on it though. If I have the Siproxd enabled, does it act as a one single server that connects multiple times to my provider or providers and then I connect to the Siproxd in return? Or, I can still register from Asterisk directly with the provider(s) and Siproxd will take care of the SIP packets to be handled nicely? If it's the latter then it sounds fine to use otherwise it would not only be complicated but also a downtime to Siproxd mean downtime to all Asterisk servers. ***In addition I have setup Siproxd according to pfsense guide online but once I save the configurations and return to it there are no configs left. I know this question is for pfsense forum but maybe someone else experienced this? ***And to return to my original question, do I need a SIP proxy and which one would be suit my needs? I still like to get an input on my previous e-mail. I have to stay with pfsense for now as it has proven to be a good router in all other aspect. Thanks, On Tue, Jan 11, 2011 at 7:38 AM, Pan B. Christensen p...@ibidium.no wrote: Hello Bruce, Your understanding of NAT is correct, and your setup should work. I’m not familiar with Pfsense, but I suspected that your problem was due to a SIP ALG. Pfsense seems to have a SIP ALG and other special handling of VoIP traffic. Hence, you are not using plain NAT. Pfsense is probably rewriting the SIP packets in addition to the IP packets. Try reconfiguring Pfsense or swapping it for something else. A good way to troubleshoot your scenario is to compare the traffic in your end to the traffic on your providers end (or on either side of pfsense). Pay attention to the source and destination IP and ports in addition to the contents of the SIP messages. http://doc.pfsense.org/index.php/VoIP_Configuration http://en.wikipedia.org/wiki/Application-level_gateway With kind regards, Pan *From:* Bruce B bruceb...@gmail.com *Sent:* Tuesday, January 11, 2011 8:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Subject:* [asterisk-users] Do I need a sip proxy? Hi Everyone, I am running multiple instances of Asterisk in Proxmox and so far I had one central Asterisk feeding all others with trunks from one provider. Now, I want to connect each Asterisk server directly to the provider. Based on my understanding, each connection made to the provider port 5060 would be on a port that is unique to that server. And so other connections made to the same provider will go out through a different port and should receive responses through that different port. At least that is my understanding of NAT. The provider should see me trying to register from the same IP with multiple different ports (high number ports; not talking about 5060 as this is outbound and not inbound) and should be able to differentiate between SIP packets coming from various servers. However, it seems to not happen. There is some sort of clash and only one of the servers shows registered with the provider and other's trunks go down. I have noticed that keeping one server works. It could also be that my Fail2ban kicks in on all servers if the SIP packets received are broadcasted to all servers which shouldn't really happen and router should take of this by sending it to the server that has the established connection through that port. *My equipment:* Asterisk 1.6x Pfsense 1.2.3 Dumb Switch *My questions:* A- What is the rational behind this? B- Do I need a sip proxy server? Something like Siproxd, OpenSIPs, or Kamailio? C- Which one of the above is the easiest to get running given I never tried any of those. D- If I am doing an SIP proxy server then it might have to also be redundant. What options do I have in that and which of above or any other suggested package might be great for future expansions. Clarification on how NAT would work in situations like this would be much appreciated. Thanks -- -- _ -- 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
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
[asterisk-users] Show voicemail in GUI
Hello list, I have a management user interface written in php for controlling some functions of Asterisk PBX. I use realtime a lot. Is there a way to easily get the details of a voicemail account and the messages that have been left ? In use realtime voicemail, but how to get the messages that have been left for a certain mailbox-extension ? Kind regards, Jonas. -- _ -- 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 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] Show voicemail in GUI
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 11, 2011 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Show voicemail in GUI Hello list, I have a management user interface written in php for controlling some functions of Asterisk PBX. I use realtime a lot. Is there a way to easily get the details of a voicemail account and the messages that have been left ? In use realtime voicemail, but how to get the messages that have been left for a certain mailbox-extension ? Kind regards, Jonas. I will qualify my answer by saying that I don't mess with RealTime because I have other more pressing issues than nice features of Asterisk. From what I read, realtime controls the user information for voicemail, but the nuts and bolts information you would need for this GUI interface is still going to be in /var/spool/asterisk/voicemail/default. To see the messages for user 100, I would do ls /var/spool/asterisk/voicemail/default/100/INBOX or ls /var/spool/asterisk/voicemail/default/100/INBOX/*.txt. It is entirely possible that Realtime actually puts this in a table, but I don't know. -- _ -- 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] Call queues on load-balanced asterisks
Hi Pan Dhaval, We have implemented a FastAGI based queue with Erlang for a inbound call center, and call this new application as FlexQueue. All calls distributed on multiple asterisk boxes go through and are controlled by that same remote fastagi server. It can routing calls to any destination, by any business rules. It don't rely on the db for agent/call status store query. It's event driven and dict based agent/call store query, with very good performance, and low cpu power consumption. I think for your requirement, app_queue could not fulfill that. Best Regards, Thomas Liu - WShuttle Infotech Ltd. http://www.wshuttle.com / http://www.lookmypc.com http://www.vicidial.cn / http://www.call-center-software.com.cn Tel: +86 20 39230098 39230096 Mobile : +86 1390 3051 930 HK DID: +852 6950 0916, Macau DID: +853 6285 0645 Email: thomas@wshuttle.com MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly Yahoo Messenger: thomaslly Address: Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area, Guangzhou Higher Education Mega Center, Guangzhou, Guangdong Province, China. Zip code: 510006 -- Hello Dhaval (and others), As far as I can tell, realtime queue will not solve my problem. I can statically define the same queue with the same members on two machines as well. I was planning to use realtime anyway. The issue is the actual queueing of the incoming calls. Let?s say I define the queue IT-support with members Local/100 and Local/101 on both machines. The first call comes in and is distributed by Kamailio to Asterisk A, and answered by 100. The next call comes in to Asterisk B, and is answered by 101. At this point, both members are busy. Call 3 now comes in and is sent to Asterisk A, where it waits for a free member. Call 4 comes in and is also sent to Asterisk A, as is Call 5. Then call 6 is sent to Asterisk B. At this point 100 finishes his call and becomes free. Which call is delivered to 100? As far as I can tell, that?s a 50/50 chance between call 3 and call 6. This is not correct behaviour! Call 6 should wait until calls 3, 4 and 5 (from the other server) have all been delivered. In the example above: When call 3 comes in, Asterisk A may even try to deliver it to 101, who gets call waiting indication. He will now have two simultaneous calls from the same queue! I have not found any way to share information about calls waiting in the queue, wait times, member states and so on between the two servers. Unless you guys know of a way, I think I'm going to have to ask the customer to change their design to master-slave (with failover) instead of load-balanced. With kind regards, Pan Hello Pan, You can user DB for this just make real time configuration of Queue and make all asterisk server connected to Same DB if more load then use replication for different server on DB, also So that Quque name should be same for all server and asterisk can call same agent. you didnot mentioned that which purpose youwere use queue other wise i can give answer in better way. regards Dhaval On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen pan at ibidium.no wrote: Hello, I have been asked to implement the following design: Load-balanced Kamailio servers handling registrations and routing. Load-balanced asterisk feature servers handling voicemail and other things Kamailio cannot do. Plus several load-balanced gateways, but they are not relevant to my question. All this is working fine. I've now been asked to start implementing calling queues, and my question is this: How can I implement the same queue on multiple Asterisk servers? Let's say that 10 people call the same queue. These calls would then currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make Asterisk A respect the 5 people queued on the other server and vice versa? Will the customer need to change their design to make the feature servers master-slave with failover instead of load-balanced? Mvh Pan -- _ -- 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] Show voicemail in GUI
On Tue, 2011-01-11 at 16:52 +0100, Jonas Kellens wrote: Hello list, I have a management user interface written in php for controlling some functions of Asterisk PBX. I use realtime a lot. Is there a way to easily get the details of a voicemail account and the messages that have been left ? In use realtime voicemail, but how to get the messages that have been left for a certain mailbox-extension ? Kind regards, Jonas. -- You can see how many messages, Old and new are in any mailbox using the AMI http://www.voip-info.org/wiki/view/Asterisk+manager+API Then you can use the command MailboxCount to find out how many messages are in the mailbox Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 ?
Why not an unattended transfer to the queue itself, or a different queue? l. 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
[asterisk-users] Asterisk hardware server
Hi All, I am planing to implement asterisk server but i have confusion regarding which hardware should i pick ? We have standard IBM servers in data center so i am planing to pick IBM x3550. so just wanted to know whether sangoma PRI card is supported with this server hardware. anyone configured PRI card on IBM servers ? Thanks, 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
Re: [asterisk-users] Asterisk hardware server
On Tue, Jan 11, 2011 at 2:16 PM, satish patel satish...@hotmail.com wrote: Hi All, I am planing to implement asterisk server but i have confusion regarding which hardware should i pick ? We have standard IBM servers in data center so i am planing to pick IBM x3550. so just wanted to know whether sangoma PRI card is supported with this server hardware. anyone configured PRI card on IBM servers ? Thanks, S Satish With the new PCI-E cards there is no more concern over interrupts or compatibility. ~~~ Andrew lathama Latham lath...@gmail.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
Re: [asterisk-users] Asterisk hardware server
Great! so IBM x3550 would be good choice for me with PCI-E card ;) Date: Tue, 11 Jan 2011 14:29:28 -0300 From: lath...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk hardware server On Tue, Jan 11, 2011 at 2:16 PM, satish patel satish...@hotmail.com wrote: Hi All, I am planing to implement asterisk server but i have confusion regarding which hardware should i pick ? We have standard IBM servers in data center so i am planing to pick IBM x3550. so just wanted to know whether sangoma PRI card is supported with this server hardware. anyone configured PRI card on IBM servers ? Thanks, S Satish With the new PCI-E cards there is no more concern over interrupts or compatibility. ~~~ Andrew lathama Latham lath...@gmail.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 reject an incoming call using AMI ?
2011/1/11 Lenz Emilitri lenz.lo...@gmail.com Why not an unattended transfer to the queue itself, or a different queue? l. I was afraid that the incoming call could be presented to the same agent again. Thinking back about it, it seems to fit what I'm after. Thanks ! -- _ -- 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] Asterisk hardware server
On Tue, Jan 11, 2011 at 2:47 PM, satish patel satish...@hotmail.com wrote: Great! so IBM x3550 would be good choice for me with PCI-E card ;) There are many models in that series but a quick look shows that they are all PCI-E compatible. You should have no trouble. ~~~ Andrew lathama Latham lath...@gmail.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
Re: [asterisk-users] How to reject an incoming call using AMI ?
On 01/11/2011 11:50 AM, Olivier wrote: 2011/1/11 Lenz Emilitri lenz.lo...@gmail.com mailto:lenz.lo...@gmail.com Why not an unattended transfer to the queue itself, or a different queue? l. I was afraid that the incoming call could be presented to the same agent again. Thinking back about it, it seems to fit what I'm after. If there is a call leg out to the agent that is in 'ringing' state, then issuing a Hangup on that call leg is not going to affect the call sitting in the queue waiting for an agent to answer... they are not connected to each other until the agent answers and the queue application connects them together. Issuing an AMI Hangup on the call leg that is ringing the agent's phone is exactly what you want to do. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Asterisk timezone issue
Hi All, We are planing to centralized our asterisk for all sites but now question is timezone, we have one site at California PST time zone and other site at Boston EST timezone. Now question is if i put central asterisk at California in PST time. how could my all AGI and other time related application for for EST timezone ? because there is 3 hrs time difference. Is there any work around for this ? Thanks, 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
Re: [asterisk-users] Asterisk timezone issue
On Tue, Jan 11, 2011 at 3:31 PM, satish patel satish...@hotmail.com wrote: Hi All, We are planing to centralized our asterisk for all sites but now question is timezone, we have one site at California PST time zone and other site at Boston EST timezone. Now question is if i put central asterisk at California in PST time. how could my all AGI and other time related application for for EST timezone ? because there is 3 hrs time difference. Is there any work around for this ? Thanks, S There are actually many ways to handle this. The most transparent method would be by context. One context per location and updating the language and timezone per context. Beware of the res_phoneprov using only the server TZ config. ~~~ Andrew lathama Latham lath...@gmail.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
Re: [asterisk-users] asterisk fax problem
Hello, Thanks a lot Kevin and Steve.. That really makes sense. The version 1.6.2.9-2 was the version on debians own repository. Not a version I compiled. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, January 11, 2011 4:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk fax problem On 01/11/2011 06:48 AM, Steve Underwood wrote: On 01/11/2011 04:14 PM, Oguzhan Kayhan wrote: Hello, I have asterisk 1.6.2.9-2 I tried to install fax utility as it is shown on pdf documents on asterisk site. I downloaded Opteron compiled res_fax and res_fax_digium files and copied to /usr/lib/asterisk/modules/ where default modules directory is. I created a free fax license and created license file on asterisk server too. WHen i run asterisk it crashed. I noticed that if res_fax.so file is in directory (even with _digium file or not) i got error already have an application 'RECEIVE Fax' and 'SEND Fax' If i remove res_fax.so file from directory asterisk loads but i got the following error msgs on debug [Jan 11 09:07:03] WARNING[11898]: loader.c:435 load_dynamic_module: Error loading module 'res_fax_digium': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_register [Jan 11 09:07:03] WARNING[11898]: loader.c:801 load_resource: Module 'res_fax_digium' could not be loaded. Cam sonebody help me about what is going on :) Install spandsp as your FAX engine, and make your problems go away. He already has it, he has app_fax loaded by default (by whatever Asterisk packages he is using, since 1.6.2.9-2 is not an official Asterisk version number). Oguzhan: Your Asterisk installation already has SendFAX and ReceiveFAX applications (based on Steve's excellent spandsp library) installed, and you can use them right now. If you want to try Digium's Fax For Asterisk modules, you'll have to add noload = app_fax.so to your /etc/asterisk/modules.conf file, so that you don't have two modules trying to provide the same SendFAX/ReceiveFAX applications. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Using the Telco Call Transfer Features.
Does anyone have any tips on how to configure asterisk to use the flash and dial codes that my telco provides for transferring calls to outside lines? -- _ -- 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] Using the Telco Call Transfer Features.
Jeff B wrote: Does anyone have any tips on how to configure asterisk to use the flash and dial codes that my telco provides for transferring calls to outside lines? I don't know how it works, but here is the wiki page on it: http://www.voip-info.org/wiki/view/Asterisk+cmd+Flash Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Asterisk timezone issue
Thank you so much, We have separate server for phone provisioning. so look like no issue there. Thanks, S Date: Tue, 11 Jan 2011 15:36:35 -0300 From: lath...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk timezone issue On Tue, Jan 11, 2011 at 3:31 PM, satish patel satish...@hotmail.com wrote: Hi All, We are planing to centralized our asterisk for all sites but now question is timezone, we have one site at California PST time zone and other site at Boston EST timezone. Now question is if i put central asterisk at California in PST time. how could my all AGI and other time related application for for EST timezone ? because there is 3 hrs time difference. Is there any work around for this ? Thanks, S There are actually many ways to handle this. The most transparent method would be by context. One context per location and updating the language and timezone per context. Beware of the res_phoneprov using only the server TZ config. ~~~ Andrew lathama Latham lath...@gmail.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
[asterisk-users] Issue with Red Alarm with DAhDi
Hi! I have an analog line connected to my asterisk and when I try to answer a call I get this -- Starting simple switch on 'DAHDI/7-1'-- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack-- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack-- DAHDI/7-1 Playing 'vm-intro' (language 'en')[Jan 11 16:29:46] WARNING[3411]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1'-- Hungup 'DAHDI/7-1'[Jan 11 16:29:47] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7-- Starting simple switch on 'DAHDI/7-1'-- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack-- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack-- DAHDI/7-1 Playing 'vm-intro' (language 'en')[Jan 11 16:29:52] WARNING[3412]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1'-- Hungup 'DAHDI/7-1'[Jan 11 16:29:53] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7-- Starting simple switch on 'DAHDI/7-1'-- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack-- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en')[Jan 11 16:29:58] WARNING[3413]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' I checked fisically the card and not red alarm in this. I am using Asterisk 1.4.38 and Dahdi 2.4.0 Any cluees ? TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- 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] Using the Telco Call Transfer Features.
Doug, Thanks that sounds a lot like what i was looking for. On Tue, Jan 11, 2011 at 2:24 PM, Doug Lytle supp...@drdos.info wrote: Jeff B wrote: Does anyone have any tips on how to configure asterisk to use the flash and dial codes that my telco provides for transferring calls to outside lines? I don't know how it works, but here is the wiki page on it: http://www.voip-info.org/wiki/view/Asterisk+cmd+Flash Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] OpenVPN + SIP configuration?
I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting through Ethernet in hotels or public wifi hotspots. I use OpenVPN to pass both IAX2 trunking and SIP for a softphone on a laptop. Works very well. If the OpenVPN server is on a uClinux appliance, my first question is does the * server know how to route to the OpenVPN client address? If the appliance is not the default gateway for the server, a route to the OpenVPN client via the uClinux appliance address needs to be added. Since the client is Windows, you do have to start the client (GUI) as 'Administrator' otherwise the routing table does not get updated and the client cannot route to the server network. Dale -- _ -- 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] Do I need a sip proxy?
Hi, At least that is my understanding of NAT. The provider should see me trying to register from the same IP with multiple different ports (high number ports; not talking about 5060 as this is outbound and not inbound) and should be able to differentiate between SIP packets coming from various servers. However, it seems to not happen. There is some sort of clash and only one of the servers shows registered with the provider and other's trunks go down. I have noticed that keeping one server works. What I have noticed with consumer grade NAT routers is that they seem to be optimized to only keep track of one single client that is allowed to connect to a server:port tuple on the outside. So if a SIP client on local ip_a and port 5060 on the inside of the router is talking to a server outside of the NAT at ip_s and port 5060 it works fine, but the minute a second client at local IP ip_b and port 5060 starts to talk to ip_s at port 5060 on the outside of the same NAT router all traffic from server_s is sent to ip_b and ip_a will receive nothing. With NAT entry timeouts and regular traffic from ip_a and ip_b you might see only one local client being reachable all the time or connectivity hopping from one to te other at regular intervals. There are NAT implementations that do not have this problem, but that might require a more expensive router or you can configure the SIP clients to all use different local ports. Perhaps this is a result of some sort of SIP ALG or a stupid basic NAT implementation to reduce code complexity on the router, but it is a nuisance either way. -- Andreas Sikkema -- _ -- 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] Unable to get Fax t38 working with IrisTel trunk
Hi everyone, I have been trying to get T.38 Faxing to work with Iristel sip trunks for last few days but havn't been sccussful. I am using Asterisk 1.6.2.8 and SpanDSP 0.6. Here is what I see in the tcpdump capture: 1. Call come in from the trunk as regular voice call with g.711 codec 2. Asterisk answers the call and recognizes the CNG and sends the call to fax extension 3. Eventually Receive fax is called with a file name 4. Asterisk sends update message to remote gateway with T38 codec information 5. Remote server doesn't respond. Asterisk resends update messages multiple times. 6. Eventually remote gateway sends the invite with T38 codecs listed in the SDP 7. Asterisk Responds back with 488 Not acceptable here 8. Another invite is send from the remote gateway with T38 codecs in the SDP 9. Asterisk sends OK but g.711 codecs listed in SDP. 10. remote gateway sends the BYE message and call is completed. The questions I have are: Why Asterisk sends update message in step 4 above instead of send an invite? Why Asterisk responds back with 488 in step 7 above? Why asterisk sends g.711 codecs in step 9 above? Thanks Karim -- _ -- 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] Issue with Red Alarm with DAhDi
On 1/11/11 2:33 PM, Edwin Quijada wrote: Hi! I have an analog line connected to my asterisk and when I try to answer a call I get this -- Starting simple switch on 'DAHDI/7-1' -- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en') [Jan 11 16:29:46] WARNING[3411]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' [Jan 11 16:29:47] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7 -- Starting simple switch on 'DAHDI/7-1' -- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en') [Jan 11 16:29:52] WARNING[3412]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' [Jan 11 16:29:53] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7 -- Starting simple switch on 'DAHDI/7-1' -- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en') [Jan 11 16:29:58] WARNING[3413]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' I checked fisically the card and not red alarm in this. I am using Asterisk 1.4.38 and Dahdi 2.4.0 Any cluees ? TIA What card are you using for your DAHDI channels? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 ?
2011/1/11 Kevin P. Fleming kpflem...@digium.com On 01/11/2011 11:50 AM, Olivier wrote: 2011/1/11 Lenz Emilitri lenz.lo...@gmail.com mailto: lenz.lo...@gmail.com Why not an unattended transfer to the queue itself, or a different queue? l. I was afraid that the incoming call could be presented to the same agent again. Thinking back about it, it seems to fit what I'm after. If there is a call leg out to the agent that is in 'ringing' state, then issuing a Hangup on that call leg is not going to affect the call sitting in the queue waiting for an agent to answer... they are not connected to each other until the agent answers and the queue application connects them together. Issuing an AMI Hangup on the call leg that is ringing the agent's phone is exactly what you want to do. Thanks for clarifying this ! Cheers -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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
