I am trying to set add a SIP Header to a call before adding it to the Queue.
The dial plan sends the call to my macro to perform the work. When I use
chan_sip, everything works as expected. When I use PSJIP support, it's not
adding the SIP header.
Looking at the output, I see the macro is called in both cases. In the PJSIP
case, the added sip header never is showing up in the asterisk logs (verbose
999). In the SIP case, I see it.
Does the function Set(PJSIP_HEADER(add, ..... not transfer over to the call when
the Queue function is called?
Am I calling the Set(PJSIP_Header(add portion incorrectly? Or is this a problem
with the Asterisk PJSIP support?
chan_sip...
[macro-MY-SetDNID]
exten => s,1,Verbose(X-MY-DNID:${MY_DNID})
exten => s,1,SIPAddHeader(X-MY-DNID:${MY_DNID})
same => n,Queue(${ARG2})
pjsip....
[macro-MY-SetDNID]
exten => s,1,Verbose(X-MY-DNID:${MY_DNID})
same => n,Set(PJSIP_HEADER(add,X-MY-DNID)=${MY_DNID})
same => n,Queue(${ARG2})
Have a great day!
Dan
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Dan Cropp wrote:> I am trying to set add a SIP Header to a call before adding it to the Queue. > > The dial plan sends the call to my macro to perform the work. When I use > chan_sip, everything works as expected. When I use PSJIP support, it?s > not adding the SIP header. > > Looking at the output, I see the macro is called in both cases. In the > PJSIP case, the added sip header never is showing up in the asterisk > logs (verbose 999). In the SIP case, I see it. > > Does the function Set(PJSIP_HEADER(add, ?.. not transfer over to the > call when the Queue function is called? > > Am I calling the Set(PJSIP_Header(add portion incorrectly? Or is this a > problem with the Asterisk PJSIP support?PJSIP_HEADER works on the channel it is invoked on. SIPAddHeader does things differently and uses channel variables underneath which can be inherited. What is the exact call flow and where do you expect the headers to appear? -- 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
In doing a little research, it seems the Referred-By header could be added after
the pjsip_xfer_initiate.
This is the approach PJSIP did for some code as far back as PJSIP 1.6.
/*
* Create REFER request.
*/
status = pjsip_xfer_initiate(sub, dest, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create REFER request", status);
pjsip_dlg_dec_lock(dlg);
return status;
}
/* Add Referred-By header */
gs_hdr = pjsip_generic_string_hdr_create(tdata->pool, &str_ref_by,
&dlg->local.info_str);
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)gs_hdr);
/* Add additional headers etc */
pjsua_process_msg_data( tdata, msg_data);
/* Send. */
status = pjsip_xfer_send_request(sub, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send REFER request", status);
pjsip_dlg_dec_lock(dlg);
return status;
}
Could anyone provider some insight into how difficult this might be for me to
add and submit for approval? Depending on the answer, my manager may be willing
to let me work on this.
I've developed in C/C++ for over 25 years so I'm plenty familiar with
the language.
I'm less familiar with the syntax and coding standards of Asterisk. I know
the group is very good at letting people know about their mistakes and how to
fix them.
Have a great day!
Dan
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Joshua Colp
Sent: Tuesday, August 25, 2015 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP add
Dan Cropp wrote:> I am trying to set add a SIP Header to a call before adding it to the
Queue.
>
> The dial plan sends the call to my macro to perform the work. When I
> use chan_sip, everything works as expected. When I use PSJIP support,
> it's not adding the SIP header.
>
> Looking at the output, I see the macro is called in both cases. In the
> PJSIP case, the added sip header never is showing up in the asterisk
> logs (verbose 999). In the SIP case, I see it.
>
> Does the function Set(PJSIP_HEADER(add, ..... not transfer over to the
> call when the Queue function is called?
>
> Am I calling the Set(PJSIP_Header(add portion incorrectly? Or is this
> a problem with the Asterisk PJSIP support?
PJSIP_HEADER works on the channel it is invoked on. SIPAddHeader does things
differently and uses channel variables underneath which can be inherited. What
is the exact call flow and where do you expect the headers to appear?
--
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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Thank you for responding Joshua.
My goal is to send a custom header in the INVITE packet.
A developer wrote a macro using Asterisk and the SIPAddHeader many years ago.
He has custom software that looks for this custom header field to know what
number the original call came into Asterisk on.
After the SIPAddHeader, a Queue is called.
The basic call flow is the call comes in,
It copies the EXTEN to a DNID variable
exten => _X.,n,Set(__MY_DNID=${EXTEN})
It is sent to AsyncAGI.
exten => _X.,n,AGI(agi:async)
Using AMI, we control the call from there.>From AMI, we Answer the call.
For testing the very next thing I do is
Action: AGI
ActionID: C15
Channel: PJSIP/3400-00000000
CommandID: C15
Command: EXEC Macro My-SetDNID,${MY_DNID},TestApp
[macro-My-SetDNID]
exten => s,1,Verbose(X-My-DNID:${MY_DNID})
; SIP version (which works)
same => n,SIPAddHeader(X-My-DNID:${MY_DNID})
; PJSIP version code
; same => n,Set(PJSIP_HEADER(add,X-My-DNID)=${MY_DNID})
same => n,Queue(${ARG2})
When using chan_sip, I'm seeing the following INVITE sent to the Agent in
the queue...
18:48:01.650180 IP (tos 0x0, ttl 64, id 44633, offset 0, flags [none], proto UDP
(17), length 950)
192.168.xxx.xxx.sip > 192.168.yyy.yyy.5063: SIP, length: 922
INVITE sip:TestApp_EmergencyAgent1 at 192.168.yyy.yyy:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK77caff71
Max-Forwards: 70
From: <sip:3400 at 192.168.xxx.xxx>;tag=as13bb4901
To: <sip:TestApp_EmergencyAgent1 at 192.168.yyy.yyy:5063>
Contact: <sip:3400 at 192.168.xxx.xxx:5060>
Call-ID: 32d64cd37cc3673661a5819818e6201f at 192.168.xxx.xxx:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.5.0
Date: Tue, 25 Aug 2015 18:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-My-DNID: 3344
Content-Type: application/sdp
Content-Length: 290
v=0
o=root 1563523699 1563523699 IN IP4 192.168.xxx.xxx
s=Asterisk PBX 13.5.0
c=IN IP4 192.168.xxx.xxx
t=0 0
m=audio 17706 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
When using PJSIP, the header
X-My-DNID: 3344 is missing
When using PJSIP, is it possible to have an additional header added into the
INVITE when the Asterisk Queue method is called?
Have a great day!
Dan
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Joshua Colp
Sent: Tuesday, August 25, 2015 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP add
Dan Cropp wrote:> I am trying to set add a SIP Header to a call before adding it to the
Queue.
>
> The dial plan sends the call to my macro to perform the work. When I
> use chan_sip, everything works as expected. When I use PSJIP support,
> it's not adding the SIP header.
>
> Looking at the output, I see the macro is called in both cases. In the
> PJSIP case, the added sip header never is showing up in the asterisk
> logs (verbose 999). In the SIP case, I see it.
>
> Does the function Set(PJSIP_HEADER(add, ..... not transfer over to the
> call when the Queue function is called?
>
> Am I calling the Set(PJSIP_Header(add portion incorrectly? Or is this
> a problem with the Asterisk PJSIP support?
PJSIP_HEADER works on the channel it is invoked on. SIPAddHeader does things
differently and uses channel variables underneath which can be inherited. What
is the exact call flow and where do you expect the headers to appear?
--
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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Possibly Parallel Threads
- Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?
- Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?
- PJSIP_HEADER - Diversion header manipulation
- PJSIP: how to retrieve underlying SIP Call-ID
- Can I use PJSIP_HEADER to read the SIP 183 message header?