[asterisk-dev] SIP To: header

sip sip at arcdiv.com
Tue Jul 11 14:25:15 MST 2006


Is there a way to access the actual SIP To: header?  I know the URI is easily
accessible, and is handy for a multitude of things, but in a scenario in which
a call has been forwarded from one URI to another, it's handy to know whence
the forward was initiated (which would only be in the To: header presumably). 

N.



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