[asterisk-users] PJSIP add header on forwarded call

[Joshua C. Colp]

On Tue, Nov 27, 2018, at 8:13 AM, Administrator TOOTAI wrote:
> Le 27/11/2018 à 12:13, Joshua C. Colp a écrit :
> > On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...]
> >>
> >> [TOOTAiAudio]
> >> ;
> >> ; Call our gateway
> >>
> >> exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1})
> >>    same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T)
> >>    same = n,Return
> >>
> >> exten = h,1,NoOp()
> >>    same = n,NoOp(Hangup Cause: ${HANGUPCAUSE})
> >>    same = n,NoOp(Dial status : ${DIALSTATUS})
> >>    same = n,NoOp(X-TOOTAiAudio=${PJSIP_HEADER(read,X-TOOTAiAudio-CALLED)})
> >>    same = n,Return
> [...]
> >>
> >> Why can't be PJSIP extra headers setted in this case ?
> > 
> > As documented on the wiki[1] the PJSIP_HEADER dialplan function has to be executed on the PJSIP channel itself, not the calling channel. You need to use a pre-dial handler and invoke it there.
> > 
> > [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER
> > 
> 
> Thanks Joshua, that worked. As you see above I want to have the value of 
> headers when call is ended. Problem is that on h extension the channel 
> already gone.
> 
> Is there a solution to archieve this ?

Is there a reason you can't use a normal dialplan variable instead? Otherwise I don't believe PJSIP_HEADER will retrieve such information regardless, it's for querying headers on an incoming INVITE.

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org