[asterisk-users] Pass DTMF to IVR gateway through SIP phone conferencing.

[C. Savinovich]

Without reading too much into your description, I can tell you that being an
inband sound, and as long as the dtmf tone is "heard" by everybody during the
conference, and being the ivr gateway one of the parties of the conference, I
don't see a reason why the ivr gateway wouldn't act upon hearing the dtmf tone. 
It wouldn't know who pressed it, although if that matters, can be arranged by
writing a patch to the meetme application where you can identify the channel
that pressed the dtmf tone.

Best
Chris Savinovich

On December 20, 2010 at 6:56 AM Asterisk Man <theasteriskman at gmail.com> wrote:


> Will someone help/direct me find a way to implement this?
> Or you can suggest some other method.
> 
> 
> On Fri, Dec 17, 2010 at 12:44 PM, Asterisk Man <theasteriskman at gmail.com
> [mailto:theasteriskman at gmail.com] >wrote:
> 
> > Hi friends,
> > 
> > I want to implement following scenario using Asterisk. Please suggest me
> > whether it is possible  or
> > 
> > not.
> > 
> > This is bit off Asterisk and more on SIP side.
> > 
> > An Asterisk box with one Station(SIP channel) and PRI.
> > 
> > Agent dials a PSTN number of customer from station through Asterisk PRI.
> > Agent gets connected with
> > 
> > customer. Agent puts customer on hold. Agent dials another PSTN number which
> > is of IVR gateway.
> > 
> > Agent now makes conference(Station facility)  with customer and IVR gateway.
> > Gateway plays an IVR
> > 
> > asking customer to enter his customer id number.
> > 
> > My question is, will DTMF get forwarded to IVR gateway?
> > 
> > I am asked to implement this and not having PRI for the moment in my
> > Asterisk box.
> > 
> > Thanking you in advance.
> > 
> > -AsteriskMan
> > 

> 



Christian Savinovich
Telecom & Telephony Consulting
646.982.3572
c.savinovich at itntelecom.com
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