Re: [asterisk-users] PJSIP add header on forwarded call
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
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Re: [asterisk-users] PJSIP add header on forwarded call
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
--
_
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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
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Re: [asterisk-users] PJSIP add header on forwarded call
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 ?
--
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
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Re: [asterisk-users] PJSIP add header on forwarded call
On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:
> Hi list,
>
> to manage an external queue agent the only solution I found is to
> connect a local account and redirect calls to this account using forward
> features from the phone (SNOM). The problem I face is that before
> calling the agent I would like to set extra header. Dialplan to call
> external agent is this one with (Gosub):
>
> [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
>
> When a local phone call extension 115 (the one where calls to external
> agent are forwarded), everything is working well. But if I call the
> account from a queue I get
>
>
> [Nov 27 09:54:08] ERROR[12758][C-005f]: res_pjsip_header_funcs.c:513
> func_write_header: This function requires a PJSIP channel
>
> Output of queue is
>
> deblix9*CLI> queue show q301
> q301 has 0 calls (max unlimited) in 'ringall' strategy (6s holdtime, 47s
> talktime), W:0, C:25, A:3, SL:100.0%, SL2:100.0% within 60s
> Members:
> PJSIP/PPermis115 (ringinuse disabled) (dynamic) (Not in use) has
> taken 8 calls (last was 706 secs ago)
> No Callers
>
> where PPermis115 is the local account on a phone who forward calls to
> extension 115.
>
> 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
--
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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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
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[asterisk-users] PJSIP add header on forwarded call
Hi list,
to manage an external queue agent the only solution I found is to
connect a local account and redirect calls to this account using forward
features from the phone (SNOM). The problem I face is that before
calling the agent I would like to set extra header. Dialplan to call
external agent is this one with (Gosub):
[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
When a local phone call extension 115 (the one where calls to external
agent are forwarded), everything is working well. But if I call the
account from a queue I get
[Nov 27 09:54:08] ERROR[12758][C-005f]: res_pjsip_header_funcs.c:513
func_write_header: This function requires a PJSIP channel
Output of queue is
deblix9*CLI> queue show q301
q301 has 0 calls (max unlimited) in 'ringall' strategy (6s holdtime, 47s
talktime), W:0, C:25, A:3, SL:100.0%, SL2:100.0% within 60s
Members:
PJSIP/PPermis115 (ringinuse disabled) (dynamic) (Not in use) has
taken 8 calls (last was 706 secs ago)
No Callers
where PPermis115 is the local account on a phone who forward calls to
extension 115.
Why can't be PJSIP extra headers setted in this case ?
Thanks for any hint
Reagrds
--
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
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