Le 27/11/2018 à 13:18, Joshua C. Colp a écrit :
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?
That's what I do at this time. I thought I could bypass this by
retriving the output of headers
Otherwise I don't believe PJSIP_HEADER will retrieve such information
regardless, it's for querying headers on an incoming INVITE.
Ok, thanks for your help.
--
Daniel
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astricon is coming up October 9-11! Signup is available at:
https://www.asterisk.org/community/astricon-user-conference
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
