[asterisk-users] issue with speech in IVR
salah.elharit200 at gmail.com
Fri Nov 29 02:05:16 CST 2013
yes if imake an extension-to-extension call, i can send DTMF, Both ways ====
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 .
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:
> 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.)
> Answers come *after* questions.
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