[asterisk-users] DTMF Testing software to test IVR system

Sammy Govind govoiper at gmail.com
Thu Dec 29 03:31:55 CST 2011 

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 <virbhati at gmail.com> 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 <govoiper at gmail.com> 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 <virbhati at gmail.com>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 <pabelanger at digium.com>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
>>>>
>>>>
>>>> --
>>>> ______________________________**______________________________**
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>>>
>>>
>>>
>>> --
>>>
>>> Thanks and regards
>>>
>>>  Virendra Bhati
>>> +91-8885268942
>>> Software Engineer
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
>
> --
>
> 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
>
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