[asterisk-dev] limit sip call transfer after refer message

[nicolo']

a question, on asterisk you can limit a consultation call transfer on
bridge channels?
I've tried using a particular context set with the variable
TRANSFER_CONTEXT but
this only works with blind transfers ..

On asterisk 1.8.16,chan_sip.c function get_refer_info, I forced to use
TRANSFER_CONTEXT but with no results..

other ideas?


-- 
M A Z
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