[asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

[Jonas Kellens]

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>