[asterisk-users] SIP To: header

[Steve Langstaff]

Isnt SIP_HEADER(TO) enough?

e.g.

exten => 1111,1,Answer
exten => 1111,2,Set(TO_HEADER=${SIP_HEADER(TO)})
exten => 1111,3,NoOp(TO_HEADER)
exten => 1111,4,Hangup

http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header


> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of sip
> Sent: 13 July 2006 12:12
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] SIP To: header
> 
> 
> 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).
> Ideally, I need this via AGI, but if it can be accessed 
> anywhere at all, I can
> code something up.
> 
> N.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>