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 <[email protected]>wrote: > > 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 : * > [email protected]* > Thanks and best regard > Phuong > > On Tue, Jan 11, 2011 at 2:05 AM, Phuong Hoang <[email protected] > > wrote: > >> >> >>> On Tue, Jan 11, 2011 at 1:18 AM, DHAVAL INDRODIYA < >>> [email protected]> 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 < >>>> [email protected]> 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, >>>>> 30000); >>>>> 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 : [email protected] >>>>> >>>>> Thanks and best regards. >>>>> Phuong >>>>> >>>>> On Mon, Jan 10, 2011 at 10:48 PM, DHAVAL INDRODIYA < >>>>> [email protected]> 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 < >>>>>> [email protected]> 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, 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 >>>>>>> <[email protected]>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:[email protected] <[email protected]> >>>>>>>> >>>>>>>> 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 >>>>>>>> <[email protected]>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:[email protected] <[email protected]> >>>>>>>>> >>>>>>>>> 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 < >>>>>>>>> [email protected]> 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 < >>>>>>>>>> [email protected]> wrote: >>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> >>>>>>>> _____________________________________________________________________ >>>>>>>> -- 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 >>>> >>> >>> >> >> >
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