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
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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
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
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On Mon, Oct 2, 2017, at 12:06 PM, Andre Gronwald wrote:> 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 3xxxxxThe PJSIP_HEADER dialplan function operates on the channel it is invoked on. In this case you are using it on the caller, not the called party. The wiki documentation[1] includes an example of how to apply it to an outgoing call. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
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 at 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" <andregronwald78 at gmail.com>
Sent: Monday, October 2, 2017 11:07 AM
To: "asterisk-users" <asterisk-users at 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 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-d05b16b9
e4c3
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
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