[Asterisk-Dev] Weird DTMF problem in ast_bridge_call

[Ashish Shinde]

Hi,
    I have two asterisk boxes. What I need to do is have one of the
box send a call transfer request to the other box over an ongoing SIP
call.
I am using the "ast_dtmf_stream" to send the transfer extension.
    The problem is that if I send the extension "#XXX" the the
ast_channel_bridge function receives the "#" and returns control to
"ast_bridge_call" but after that the remaining digits XXX are LOST.
The "ast_waitfordigit" call times out. Also any other non "#" digit
sequence send using "ast_dtmf_stream" is received properly by
"ast_channel_bridge".
    The weird thing is if the DTMF digits are send using a SIP phone
they are received correctly.
    Another thing is if the "ast_waitfordigit" call is replaced by
"ast_streamfile" followed by "ast_waitstream" calls the DTMF digits
are received.
    I am new to asterisk code and really don't understand why DTMF
send from SIP phones is received correctly by "ast_bridge_call" but
not the the DTMF send by another asterisk box.

   Would really be grateful if someone could help me out.

Thanks and regards,
  - Ashish