Asterisk in the middle, DTMF codec number different between A and B

Hi there,
I’ve got a difficulty with this case:
VoIP_A —> Asterisk --> VoIP_B, Asterisk in the middle and VoIP_A and VoIP_B are not registered users
VoIP_A offers codec number 102 for DTMF:
m=audio 33354 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
Asterisk answers this call and DTMF works between VoIP_A and Asterisk.

Then Asterisk dials to VoIP_B with
In the INVITE, from Asterisk to VoIP_B, the codec number for DTMF is 101
m=audio 14280 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Therefore, when Asterisk re-invite to both VoIP_A and VoIP_B (with “direct media”, let A and B talk to each other without Asterisk in the middle), the audio is good but DTMF doesn’t work due to the different codec number (102 for A but 101 for B).

Any suggestion how to make the DTMF codec number same for both A leg and B leg? Of any way to pass the same media payload data from A leg to B leg?

Thanks in advance.

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