[asterisk-users] PJSIP add header not working

[Loic Chabert]

Hi,

Following some new behaviour on PJSIP, adding SIP header must be done using
a subrouting.
Please find below my working configuration:




*[subroutine]exten => caller_handler,1,NoOp()same
=>n,Set(PJSIP_HEADER(add,X-CID)=${ARG1})same => n,Return()*

and then, add new parameters on Dial command:  *same
=>n,Dial(PJSIP/${EXTEN}@<peer>,,b(subroutine^caller_handler^1(${SIPCALLID})))*

The first "*b*" before parenthesis gave direction (if header must be added
to caller or callee). More information on
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER
.

Regards.

2017-10-02 17:06 GMT+02:00 Andre Gronwald <andregronwald78 at gmail.com>:

> Hi,
> I am trying to add a custom header to my calls to map several call-legs
> into a global call for viewing.
>
> For this to work I read the call-id from pjsip-channel and write it into
> X-CID:
>
> ######
>     -- Executing [s at macro-dialout-trunk-predial-hook:4]
> Set("PJSIP/10-00000006", "pjsipCallId=313530363933383438363436353930-1gh0bjceo933")
> in new stack
>     -- Executing [s at macro-dialout-trunk-predial-hook:5]
> Set("PJSIP/10-00000006", "PJSIP_HEADER(add,X-CID)=
> 313530363933383438363436353930-1gh0bjceo933") in new stack
>     -- Executing [s at macro-dialout-trunk:18] GotoIf("PJSIP/10-00000006",
> "0?bypass,1") in new stack
>     -- Executing [s at macro-dialout-trunk:19] ExecIf("PJSIP/10-00000006",
> "1?Set(CONNECTEDLINE(num,i)=0xxxxxxxxxxxxxx)") in new stack
>     -- Executing [s at macro-dialout-trunk:20] ExecIf("PJSIP/10-00000006",
> "1?Set(CONNECTEDLINE(name,i)=CID:3xxxxx)") in new stack
>     -- Executing [s at macro-dialout-trunk:21] ExecIf("PJSIP/10-00000006",
> "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3xxxxx)") in new stack
>     -- Executing [s at macro-dialout-trunk:22] GotoIf("PJSIP/10-00000006",
> "0?customtrunk") in new stack
>     -- Executing [s at macro-dialout-trunk:23] Dial("PJSIP/10-00000006",
> "PJSIP/0xxxxxxxxxxxxxx at 3xxxxx,300,T") in new stack
>     -- Called PJSIP/0xxxxxxxxxxxxxx at 3xxxxx
> <--- Transmitting SIP request (991 bytes) to UDP:217.23.24.100:5060 --->
> INVITE sip:0xxxxxxxxxxxxxx at sip.provid.er:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.253.185:15070;rport;
> branch=z9hG4bKPj453d15e0-de58-4945-8b95-d05b16b9e4c3
> From: <sip:+49xxxxxxxxxxx at sip.provid.er>
> <sip:+49xxxxxxxxxxx at sip.provid.er>;tag=080788ac-7c10-4cf3-86b3-359764ffb5a2
>
> To: <sip:0xxxxxxxxxxxxxx at sip.provid.er>
> <sip:0xxxxxxxxxxxxxx at sip.provid.er>
> Contact: <sip:+49xxxxxxxxx at 192.168.253.185:15070>
> <sip:+49xxxxxxxxx at 192.168.253.185:15070>
> Call-ID: de41b93b-51d8-44b5-9c34-f2c0928192b0
> CSeq: 1519 INVITE
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
> UPDATE, PRACK, REGISTER, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub
> Session-Expires: 1800
> Min-SE: 90
> Max-Forwards: 70
> User-Agent: FPBX-14.0.1.10(14.6.2)
> Content-Type: application/sdp
> Content-Length:   308
>
> v=0
> o=- 1719768133 1719768133 IN IP4 192.168.253.185
> s=Asterisk
> c=IN IP4 192.168.253.185
> t=0 0
> m=audio 55112 RTP/AVP 107 9 8 3 101
> a=rtpmap:107 opus/48000/2
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:20
> a=sendrecv
>
> <--- Received SIP response (559 bytes) from UDP:217.23.24.100:5060 --->
> [...]
>
> ######
>
>
>
>
> But I can't see that header anywhere in my call-legs. What am I missing?
>
>
> kind regards,
> andre
>
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-- 

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