Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

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


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
jonas.kellens at telen...
Guest
PostPosted: Tue Sep 02, 2014 3:03 am    Post subject: [asterisk-users] Custom SIP-header not present in call Aster Reply with quote
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@xx.xx.xx.98> ([email]sip:5006@xx.xx.xx.98[/email]);tag=as50c98b4c
To: <sip:0419@xx.xx.xx.238> ([email]sip:0419@xx.xx.xx.238[/email]);tag=as3c6e57b0
Call-ID: 6d1039bb22716c6e6dec69fb3e78a8d7@xx.xx.xx.98:5060 ([email]6d1039bb22716c6e6dec69fb3e78a8d7@xx.xx.xx.98:5060[/email])
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.
Back to top
steve-lists at geekint...
Guest
PostPosted: Tue Sep 02, 2014 4:35 am    Post subject: [asterisk-users] Custom SIP-header not present in call Aster Reply with quote
On 2 Sep 2014, at 09:03, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:
Quote:
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
Back to top
jonas.kellens at telen...
Guest
PostPosted: Tue Sep 02, 2014 4:38 am    Post subject: [asterisk-users] Custom SIP-header not present in call Aster Reply with quote
On 02-09-14 11:34, Steven Howes wrote:

Quote:
On 2 Sep 2014, at 09:03, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:
Quote:
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


OK.

Then how can I let another Asterisk server know the custom reason of hangup ? If it is not possible with custom SIP-header, then how ?



Regards,

Jonas.
Back to top
steve-lists at geekint...
Guest
PostPosted: Tue Sep 02, 2014 4:49 am    Post subject: [asterisk-users] Custom SIP-header not present in call Aster Reply with quote
On 2 Sep 2014, at 10:38, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:
Quote:
Then how can I let another Asterisk server know the custom reason of hangup ? If it is not possible with custom SIP-header, then how ?



As far as I know that’s going to require a source change. May not be the case with PJSIP though - not used that yet.


Steve
Back to top
asterisk_list at earth...
Guest
PostPosted: Tue Sep 02, 2014 5:08 am    Post subject: [asterisk-users] Custom SIP-header not present in call Aster Reply with quote
On Tuesday 02 Sep 2014, Jonas Kellens wrote:
Quote:
On 02-09-14 11:34, Steven Howes wrote:
Quote:
On 2 Sep 2014, at 09:03, Jonas Kellens <jonas.kellens@telenet.be

<mailto:jonas.kellens@telenet.be>> wrote:
Quote:
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

OK.

Then how can I let another Asterisk server know the custom reason of
hangup ? If it is not possible with custom SIP-header, then how ?

Fire off an AGI script which will (somehow) send the necessary message to the
other Asterisk server.

--
AJS

Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

--
_____________________________________________________________________
-- 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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services