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

Sammy Govind govoiper at gmail.com
Thu Dec 29 03:03:30 CST 2011


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 --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
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