Andre
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.
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.
It works well. You can also set variables with the _ option to trigger
which headers you want to add..
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.
Exp
[OutboundDial]
exten => _XXXXXXXXXX,1,NoOp(Dial Exp)
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
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
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
exten =>
_XXXXXXXXXX,n,Dial(pjsip/3332224444@vendortrunk,b(AddpjsipHeaders^s^1))
[AddpjsipHeaders]
exten =>s,1,Gosubif({"$[var1setinparrent}}"="1"]?ThisHeader,1)
exten =>s,n,Gosubif({"$[var2setinparrent}}"="1"]?ThatHeader,1)
exten
=>s,n,Gosubif({"$[varAddSessionInparrent}}"="1"]?addSessionCallInfo,1)
exten => ThisHeader,1,Set(PJSIP_HEADER(add,ThisHeader)=ValueToSet)
exten => ThisHeader,n,Return()
exten => ThatHeader,1,Set(PJSIP_HEADER(add,ThatHeader)=ValuetoSet)
exten => ThatHeader,n,Return()
exten =>
addSessionCallInfo,1,Set(PJSIP_HEADER(add,Call-Info)=<sip://127.0.0.1>\;answ
er-after=0)
exten => addSessionCallInfo,n,Return()
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
----------------------------------------
From: "Andre Gronwald" <andregronwal...@gmail.com>
Sent: Monday, October 2, 2017 11:07 AM
To: "asterisk-users" <asterisk-users@lists.digium.com>
Subject: [asterisk-users] PJSIP add header not working
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@macro-dialout-trunk-predial-hook:4]
Set("PJSIP/10-00000006",
"pjsipCallId=313530363933383438363436353930-1gh0bjceo933") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:5]
Set("PJSIP/10-00000006",
"PJSIP_HEADER(add,X-CID)=313530363933383438363436353930-1gh0bjceo933") in
new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/10-00000006",
"0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/10-00000006",
"1?Set(CONNECTEDLINE(num,i)=0xxxxxxxxxxxxxx)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/10-00000006",
"1?Set(CONNECTEDLINE(name,i)=CID:3xxxxx)") in new stack
-- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/10-00000006",
"0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3xxxxx)") in new stack
-- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/10-00000006",
"0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/10-00000006",
"PJSIP/0xxxxxxxxxxxxxx@3xxxxx,300,T") in new stack
-- Called PJSIP/0xxxxxxxxxxxxxx@3xxxxx
<--- Transmitting SIP request (991 bytes) to UDP:217.23.24.100:5060 --->
INVITE sip:0xxxxxxxxxxx...@sip.provid.er:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.253.185:15070;rport;branch=z9hG4bKPj453d15e0-de58-4945-8b95-d05b16b9
e4c3
From:
<sip:+49xxxxxxxx...@sip.provid.er>;tag=080788ac-7c10-4cf3-86b3-359764ffb5a2
To: <sip:0xxxxxxxxxxx...@sip.provid.er>
Contact: <sip:+49xxxxxxxxx@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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