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. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
