[asterisk-dev] SIP To: header

sip sip at arcdiv.com
Wed Jul 12 06:24:24 MST 2006


In looking at the sip_pvt structure, I'm not even seeing a var where this data
would be kept, so I'm guessing I'm going to have to add somewhere to store it
if I want it? Or am I looking in the wrong place? 

N.


On Tue, 11 Jul 2006 17:25:15 -0400, sip wrote
> 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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