[asterisk-users] issue with speech in IVR

[Salaheddine Elharit]

hi

yes  if imake an extension-to-extension call,  i can  send DTMF, Both ways ====
yes

in my case i don't need a Hardware SIP phone or a software SIP phones

i have just a number 05xxxxxx600

when the customer call this number i stor his number in my database and i
call him later

if he press 1 for xxxxxx 1 press 2 for  yyyyyyy

i sotre his phone number and his choice in my database

for me the issue the customer he can nto wait the speech of unless xxxx and
yyyy finished .

best regards



i use a diguim card with PRI


2013/11/29 A J Stiles <asterisk_list at earthshod.co.uk>

>  On 28/11/13 15:36, Salaheddine Elharit wrote:
>
> hi
> i follow your dialplan but the issue still the same ican't stop the speech
> and go to another context
>
>  any other idea  please
>
>  best regards .
>
> It sounds as thgough the DTMF tones are not being sent in a way that
> Asterisk is seeing .....
>
> What type of telephone technology are you using?  Hardware SIP phones,
> software SIP phones, analogue phones via an FXS card, analogue phones via a
> SIP ATA?  What codec are you using?
>
> If you make an extension-to-extension call, can you send DTMF tones down
> the line?  Both ways around?  Do they decode properly?  (You can get a
> mobile phone app for this.)
>
>
>  --
>  AJS
>
>  Answers come *after* questions.
>
>
> --
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