Pass DTMF to IVR gateway through SIP phone conferencing.

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Asterisk Users 3 Comments

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

3 thoughts on - Pass DTMF to IVR gateway through SIP phone conferencing.

  • 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

    Christian Savinovich
    Telecom & Telephony Consulting
    646.982.3572

    href=”mailto:c.savinovich@itntelecom.com”>c.savinovich@itntelecom.com

  • Christian,
    Thanks for your response.
    In my case, I was asked to do it through SIP phone 3 way call functionality
    and not the Asterisk Meetme application.
    I wanted to know if any one had done something similar in past or not.
    I am short of PRI in my test environment and hence I can’t test it
    practically.
    Well, I ‘ll try to implement it using Meetme.

    Regards,

    AsteriskMan