[asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

Dan Cropp dan at amtelco.com
Tue Jan 24 11:41:30 CST 2017

Thank you Joshua.

So there is no way to retrieve header information which may come in on subsequent packages?

If not, is there any way to make an Attended Transfer following the RFC5589?

Asking because we have a hospital with a Cisco switch.  Hospital has two calls from their Cisco switch into an Asterisk box.  Operator handling the two calls and needs to transfer Call A to be connected to call B.  Can obviously be patched inside of Asterisk.  However, the hospital wants the call to be Attended Transferred. Basically, we need to send the Transfer (REFER) with the Replaces containing the call ID, From tag, and the To Tag.

I am able to gather everything needed for the REFER field and pass that to the Transfer command (via AMI), except the To tag. 

-----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, January 24, 2017 11:27 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

On Tue, Jan 24, 2017, at 01:25 PM, Dan Cropp wrote:
> I place a call into Asterisk (from SIP phone) and the To header does 
> not have a tag.  Asterisk then sends it's Trying response, still no 
> tag in the To header.  The phone then replies with OK, this time the 
> To header includes a tag.
> Is there any way to retrieve this response To header (including the 
> tag
> field) from the dial plan?
> I have tried the PJSIP-HEADER read of the To header, but it seems to 
> only have access to the initial To header.
> I even tried reading multiple layers of the To header, but it still 
> didn't retrieve the newer dialog To headers.

The PJSIP_HEADER dialplan function currently only looks at the initial message. It does not allow access to subsequent ones.

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 --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

More information about the asterisk-users mailing list