<span style="font-family: Arial, Helvetica, Sans-Serif; font-size: 12px"><div>Andre</div>
<div> </div>
<div>For this to work we have had to go to using the b() option in the dial legs for the calls that are pasting up.</div>
<div>You call a context that gets run before the calls are made on each channel. This allows you to add headers to the new pjsip channels. </div>
<div>It works well. You can also set variables with the _ option to trigger which headers you want to add..</div>
<div> </div>
<div>The example below would add "ThisHeader", "ThatHeader" and "Call-Info" to the new channel created in the dial. You could use combinations of other variables and augment these methods to meet almost any need. </div>
<div> </div>
<div>Exp</div>
<div> </div>
<div>[OutboundDial]</div>
<div>exten => _XXXXXXXXXX,1,NoOp(Dial Exp)</div>
<div>exten => _XXXXXXXXXX,n,Set(_var1setinparrent=1) ;;Set Variable so that when you call the b() option context in your dial the first header is added</div>
<div>exten => _XXXXXXXXXX,n,Set(_var2setinparrent=1) ;;Set Variable so that when you call the b() option context in your dial the second header is added</div>
<div>
<div>
<div>exten => _XXXXXXXXXX,n,Set(_varAddSessionInparrent=1) ;;Set Variable so that when you call the b() option context in your dial the second header is added</div>
</div>
</div>
<div>exten => _XXXXXXXXXX,n,Dial(pjsip/3332224444@vendortrunk,b(AddpjsipHeaders^s^1))</div>
<div> </div>
<div> </div>
<div>[AddpjsipHeaders]</div>
<div>
<div>exten =>s,1,Gosubif({"$[var1setinparrent}}"="1"]?ThisHeader,1)</div>
</div>
<div>exten =>s,n,Gosubif({"$[var2setinparrent}}"="1"]?ThatHeader,1)</div>
<div>exten =>s,n,Gosubif({"$[varAddSessionInparrent}}"="1"]?addSessionCallInfo,1)</div>
<div> </div>
<div>exten => ThisHeader,1,Set(PJSIP_HEADER(add,ThisHeader)=ValueToSet)</div>
<div>exten => ThisHeader,n,Return()</div>
<div> </div>
<div>exten => ThatHeader,1,Set(PJSIP_HEADER(add,ThatHeader)=ValuetoSet)</div>
<div>exten => ThatHeader,n,Return()</div>
<div> </div>
<div>exten => addSessionCallInfo,1,Set(PJSIP_HEADER(add,Call-Info)=<sip://127.0.0.1>\;answer-after=0)<br />
exten => addSessionCallInfo,n,Return()</div>
<div> </div>
<div>Thanks<br />
<br />
Bryant Zimmerman (ZK Tech Inc.)<br />
616-855-1030 Ext. 2003</div>
<div> </div>
<hr align="center" size="2" width="100%" />
<div><span style="font-family: tahoma,arial,sans-serif; font-size: 10pt;"><b>From</b>: "Andre Gronwald" <andregronwald78@gmail.com><br />
<b>Sent</b>: Monday, October 2, 2017 11:07 AM<br />
<b>To</b>: "asterisk-users" <asterisk-users@lists.digium.com><br />
<b>Subject</b>: [asterisk-users] PJSIP add header not working</span>
<div> </div>
<p>Hi,<br />
I am trying to add a custom header to my calls to map several call-legs into a global call for viewing.<br />
<br />
For this to work I read the call-id from pjsip-channel and write it into X-CID:<br />
<br />
######<br />
-- Executing [s@macro-dialout-trunk-predial-hook:4] Set("PJSIP/10-00000006", "pjsipCallId=313530363933383438363436353930-1gh0bjceo933") in new stack<br />
-- Executing [s@macro-dialout-trunk-predial-hook:5] Set("PJSIP/10-00000006", "PJSIP_HEADER(add,X-CID)=313530363933383438363436353930-1gh0bjceo933") in new stack<br />
-- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/10-00000006", "0?bypass,1") in new stack<br />
-- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/10-00000006", "1?Set(CONNECTEDLINE(num,i)=0xxxxxxxxxxxxxx)") in new stack<br />
-- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/10-00000006", "1?Set(CONNECTEDLINE(name,i)=<a class="moz-txt-link-freetext" href="CID:3xxxxx">CID:3xxxxx</a>)") in new stack<br />
-- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/10-00000006", "0?Set(CONNECTEDLINE(name,i)=<a class="moz-txt-link-freetext" href="CID:%28Hidden%293xxxxx%29">CID:(Hidden)3xxxxx)</a>") in new stack<br />
-- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/10-00000006", "0?customtrunk") in new stack<br />
-- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/10-00000006", "PJSIP/0xxxxxxxxxxxxxx@3xxxxx,300,T") in new stack<br />
-- Called PJSIP/0xxxxxxxxxxxxxx@3xxxxx<br />
<--- Transmitting SIP request (991 bytes) to UDP:217.23.24.100:5060 ---><br />
INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:0xxxxxxxxxxxxxx@sip.provid.er:5060">sip:0xxxxxxxxxxxxxx@sip.provid.er:5060</a> SIP/2.0<br />
Via: SIP/2.0/UDP 192.168.253.185:15070;rport;branch=z9hG4bKPj453d15e0-de58-4945-8b95-d05b16b9e4c3<br />
From: <a class="moz-txt-link-rfc2396E" href="mailto:sip:+49xxxxxxxxxxx@sip.provid.er"><sip:+49xxxxxxxxxxx@sip.provid.er></a>;tag=080788ac-7c10-4cf3-86b3-359764ffb5a2<br />
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:0xxxxxxxxxxxxxx@sip.provid.er"><sip:0xxxxxxxxxxxxxx@sip.provid.er></a><br />
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:+49xxxxxxxxx@192.168.253.185:15070"><sip:+49xxxxxxxxx@192.168.253.185:15070></a><br />
Call-ID: de41b93b-51d8-44b5-9c34-f2c0928192b0<br />
CSeq: 1519 INVITE<br />
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER<br />
Supported: 100rel, timer, replaces, norefersub<br />
Session-Expires: 1800<br />
Min-SE: 90<br />
Max-Forwards: 70<br />
User-Agent: FPBX-14.0.1.10(14.6.2)<br />
Content-Type: application/sdp<br />
Content-Length: 308<br />
<br />
v=0<br />
o=- 1719768133 1719768133 IN IP4 192.168.253.185<br />
s=Asterisk<br />
c=IN IP4 192.168.253.185<br />
t=0 0<br />
m=audio 55112 RTP/AVP 107 9 8 3 101<br />
a=rtpmap:107 opus/48000/2<br />
a=rtpmap:9 G722/8000<br />
a=rtpmap:8 PCMA/8000<br />
a=rtpmap:3 GSM/8000<br />
a=rtpmap:101 telephone-event/8000<br />
a=fmtp:101 0-16<br />
a=ptime:20<br />
a=maxptime:20<br />
a=sendrecv<br />
<br />
<--- Received SIP response (559 bytes) from UDP:217.23.24.100:5060 ---><br />
[...]<br />
<br />
######<br />
<br />
<br />
<br />
<br />
But I can't see that header anywhere in my call-legs. What am I missing?<br />
<br />
<br />
kind regards,<br />
andre</p>
</div></span>