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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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
--
_____________________________________________________________________
-- 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
