o in that case you need to observer the call flow in Server-B, i.e what is the length of sound file playing. what DTMF it requires etc etc and once you detect the call flow for a successful IVR traversal then mimic the behaviour of the call from Server-A. Thats all you can do. Think of it exactly the same as Answering Machine Detection Algorithm, but in your case its like Server-B Detection Algorithm :)
-- Regards, Sammy On Thu, Dec 29, 2011 at 2:15 PM, virendra bhati <[email protected]> wrote: > In server B if I use SendDTMF then it means I am changing programming at > server B. Actually I don't have right or permission to change programming > in server B. > > otherwise your suggestion is best for channel base communication. > > > > > On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind <[email protected]> wrote: > >> Easy, use Read() to capture the incoming DTMF from Server-B >> >> Server-A <============> Server-B >> Initiate-Call ---------------------> AnswerCall() >> SendDTMF(5)------------------> Read() >> Read()<-----------------------------SendDTMF(4) >> SendDTMF(3)------------------> Read() >> Read()<-----------------------------SendDTMF(2) >> SendDTMF(1)------------------> Read() >> >> >> Put proper GOTOIFs after reads if you like. >> >> -- >> Regards, >> Sammy >> >> On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati <[email protected]>wrote: >> >>> I originate calls from .call file and 1 channel I have at A server A and >>> another channel at B server. >>> >>> *A server code is below:-* >>> >>> exten => 43689956,1,Answer() >>> same => n,Wait(5) >>> same => n,SendDTMF(1) >>> same => n,NoOp(== ${CHANNEL(state)}==> state) >>> same => n,wait(2) >>> same => n,SendDTMF(123456789012345#) >>> same => n,NoOp(== ${CHANNEL(state)}==> state) >>> same => n,Hangup() >>> >>> _________ _________ >>> | A server | _______DTMF Send_____=> | B server | >>> |_________| <=------- Responce --------- |_________| >>> >>> *B server code is below:-* >>> At B server call come to 201 extension which is mention here.. >>> >>> exten => _20[1-6],1,Answer() >>> same => n,Ringing() >>> same => n,wait(2) >>> same => n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?* >>> AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))* >>> same => n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] || >>> $[${EXTEN}=205] || >>> $[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php)) >>> same => n,Hangup() >>> >>> Now I can send the DTMF from A to B. But How I will get the responce at >>> server A. I checked all the channels variable but they didn't reply status >>> of B server channel. All information I will get of server A. Main problem >>> is that control reach to AGI and then I don't have any rights to do any >>> update or modification on AGI. So if I can work on request and responce >>> then it will be the last solution as per my knowledge. >>> >>> Is this possible with the dialplan or I am just westing time? >>> >>> >>> >>> On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger >>> <[email protected]>wrote: >>> >>>> On 11-12-28 03:25 AM, virendra bhati wrote: >>>> >>>>> Hi list, >>>>> >>>>> Is there any way in asterisk by which I make a call from server and >>>>> then >>>>> dialplan(IVR system) gets DTMF from it. I mean to say that >>>>> automatically >>>>> DTMF is sended by channels as per user defined, >>>>> >>>>> I read there is an application sendDTMF but I don't know how we can >>>>> used it? >>>>> >>>>> like A script make the call by using localdail, .call file or any >>>>> method. >>>>> And after landing the call we send dtmf to IVR system automatically as >>>>> per >>>>> my script.. >>>>> >>>>> >>>>> *extensions.conf:-* >>>>> >>>>> >>>>> exten => 1234,1,Answer() >>>>> same => n,Read(value,**pleasePress1forSupportPress2fo** >>>>> rHelp,1,,10) >>>>> same => n,NoOp(${value}) >>>>> same => n,ExecIf($[${value}=1]?Goto(**suppot,1)) >>>>> same => n,ExecIf($[${value}=2]?Goto(**help,1)) >>>>> same => n,Hangup() >>>>> >>>>> exten=> support,1,Answer() >>>>> same => n,NoOp(you are at support section) >>>>> same => n,Hangup() >>>>> >>>>> exten=> help,1,Answer() >>>>> same => n,NoOp(you are at help section) >>>>> same => n,Hangup() >>>>> >>>>> We have DTMF based tests for the testsuite[1] that you could use. >>>> >>>> [1] >>>> http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/<http://svn.asterisk.org/svn/testsuite/asterisk/trunk/> >>>> -- >>>> Paul Belanger >>>> Digium, Inc. | Software Developer >>>> twitter: pabelanger | IRC: pabelanger (Freenode) >>>> Check us out at: http://digium.com & http://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<http://lists.digium.com/mailman/listinfo/asterisk-users> >>>> >>> >>> >>> >>> -- >>> >>> Thanks and regards >>> >>> Virendra Bhati >>> +91-8885268942 >>> Software Engineer >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >> > > > > -- > > Thanks and regards > > Virendra Bhati > +91-8885268942 > Software Engineer > > > -- > _____________________________________________________________________ > -- 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
