Call transfer with SIP INFO DTMF tones

Hi All,

I have a question about the RTP P2P mode. Is it necessary to disable p2p mode if the call transfer option is enabled. I read some data and know that if the codec of caller and callee is different, Dial command option enables “t” or ‘T’, or canreinvite setting in sip.conf is “no”, then the P2P mode is disabled.
I can understand that if the codec from both side is different, the P2P mode should be disabled. However, I think if both sides are using SIP INFO to send their DTMF tones, the transfer features should be workable in RTP P2P mode. If Asterisk receives DTMF tone, ‘#’, Asterisk may easily switch both RTP paths back to Asterisk and plays “Transfer” sounds.

I did some modifications to the source codes in Asterisk v1.4 to turn on P2P mode whether the Dial option has ‘T’ or ‘t’, and it works. However, it still has some problems in handling DTMF tones.

Is it okay, or do I miss something??

Best regards,
Adam Lee