[asterisk-users] Transfer feature - howto?

[Eric "ManxPower" Wieling]

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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