[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 : 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:5...@xx.xx.xx.98;tag=as50c98b4c To: sip:0...@xx.xx.xx.238;tag=as3c6e57b0 Call-ID: 6d1039bb22716c6e6dec69fb3e78a...@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. -- _ -- 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
Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk
On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@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-- _ -- 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
Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk
On 02-09-14 11:34, Steven Howes wrote: On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@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 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. -- _ -- 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
Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk
On 2 Sep 2014, at 10:38, Jonas Kellens jonas.kell...@telenet.be wrote: 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-- _ -- 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
Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk
On Tuesday 02 Sep 2014, Jonas Kellens wrote: On 02-09-14 11:34, Steven Howes wrote: On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@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 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
Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk
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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 : -- _ -- 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
Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk
On 02-09-14 14:22, Eric Wieling wrote: 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. Hello, I have tried sending Hangup(321) on Asterisk server B to Asterisk A but when I read HangupCause on Asterisk A it always is '21'. Good idea, but it does not seem to work. Kind regards, Jonas. -- _ -- 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
Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk
321 is not a valid Asterisk hangup cause. Valid hangupcauses are 1-127 (Q.831 cause codes) See https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, September 02, 2014 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk On 02-09-14 14:22, Eric Wieling wrote: 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. Hello, I have tried sending Hangup(321) on Asterisk server B to Asterisk A but when I read HangupCause on Asterisk A it always is '21'. Good idea, but it does not seem to work. Kind regards, Jonas. -- _ -- 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
[asterisk-users] custom sip header
hello, it is possible to include an particular sip header on outbound sip channels based on some particular conditions ? in particular I am interested to signal the context the call originated from to an on route sip proxy server. thanks, razvan radu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
