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
