Technically DTMF should be a signaling thing, but I believe Asterisk
must stay in the media stream if you want to use t/T/w/W. This may have
changed in 1.4. canreinvite=no in sip.conf would keep Asterisk in the
media stream.
Steve Glaus wrote:
Eric "ManxPower" Wieling wrote:
I don't know if this is even possible. I might be totally wrong but
once this call is on the cell network, how are you gonna communicate
with asterisk?? From what I understand, while the voice (RTP) traffic
still travels through asterisk, You have no access to any kind of
signalling. Please correct me if I'm way off base here, anyone.
You are offbase. Even with reinvites the SIP SIGNALING will continue
going thru Asterisk.
Ok. Thanks! So how does one go about getting asterisk to recognize DTMF
in this situation?
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