[asterisk-users] PJSIP add header not working

[Andre Gronwald]

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>;tag=080788ac-7c10-4cf3-86b3-359764ffb5a2
To: <sip:0xxxxxxxxxxxxxx at sip.provid.er>
Contact: <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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20171002/9eb814c4/attachment.html>