asterisk users - Sep 2014 - Custom SIP-header not present in call Asterisk to Asterisk

Hello,

I have a situation where a call comes in to my Asterisk server B. This 
call comes from another Asterisk server A. I want to tell to this server 
A why my server B hangs up.

So just before hanging up, I add a custom SIP-header :

exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten => s,n,Hangup()


But I notice that this extra SIP-header is not send within the SIP-reponse :

SIP/2.0 603 Declined
Via: SIP/2.0/UDP 
xx.xx.xx.98:5060;branch=z9hG4bK168884d7;received=xx.xx.xx.98;rport=5060
From: "5006" <sip:5006 at xx.xx.xx.98>;tag=as50c98b4c
To: <sip:0419 at xx.xx.xx.238>;tag=as3c6e57b0
Call-ID: 6d1039bb22716c6e6dec69fb3e78a8d7 at xx.xx.xx.98:5060
CSeq: 102 INVITE
Server: myasterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


How can I make this work ?


Thanks.

Jonas.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20140902/baf7de09/attachment.html>

On 2 Sep 2014, at 09:03, Jonas Kellens <jonas.kellens at telenet.be>
wrote:
> So just before hanging up, I add a custom SIP-header :
> 
> exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan)
> exten => s,n,Hangup()
SIPAddHeader only works for INVITE as far as I know.

Steve
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20140902/e4626c80/attachment.html>

Try Hangup(123) where 123 is whatever "hangup cause" you want to send
back to the caller.   The calliing Asterisk server will get the valuse back in
HANGUPCAUSE variable.

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, September 02, 2014 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Custom SIP-header not present in call Asterisk to
Asterisk

Hello,

I have a situation where a call comes in to my Asterisk server B. This call
comes from another Asterisk server A. I want to tell to this server A why my
server B hangs up.

So just before hanging up, I add a custom SIP-header :

exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten => s,n,Hangup()


But I notice that this extra SIP-header is not send within the SIP-reponse :

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20140902/2130a86c/attachment.html>