DTMF issue in SIP phone conference

I want to do the SIP phone conference in asterisk .

Caller A is speaking with B . “B” choosing the second line and calling to IVR then making the Caller A in the conference.The problem is whatever the input is given by the “B” , the IVR is able to read the DTMF. But the IVR is not able to read the DTMF which is given by the Caller A . I suspect that asterisk is not broadcasting the DTMF to all the channels which are there in the conference.

Is there any way to achieve this

MeetMe eats DMTF. ConfBridge can allow DTMF to be rebroadcast; see the dtmf_passthrough option.

I have sip phone. I want to do this using SIP Phone conference ( three way transfer ) not using asterisk application.

Conferences of the type you describe are bridged by the phone. At the moment, this is a limitation of your unnamed phones, not of Asterisk.

I am using twinkle as soft phone in my linux machine .

my extensions.conf is
exten => 777,1,Dial(SIP/A)
exten => 778,1,Dial(SIP/B)

Person c dial 777 extension number in the soft phone and speaking with Person A.Then Person A goto second line and dial 778 and speaking with Person B.Then Person A click conference button in the twinkle. Now all there are able to speak and hear.The problem is ,The person “B” is not able to receive the the Person “C” DTMF . It means asterisk is not broadcasting the DTMF in the conference.

Is there any configuration is available in asterisk to broadcast the DTMF to all channels which are there in the conference.


There is no conference, as far as Asterisk is concerned. There is a call from A to B and a call from B to C.