Re: [OpenSIPS-Users] Media server communication with OpenSIPS
Hello, The invite is relayed from OpenSIPS to Asterisk. Asterisk has UAS role here. The data i'm hoping to transfer to OPENSIPS should be sent either with the 200 OK (Asterisk to OpenSIPS), during the call, or via BYE. So neither PJSIP_HEADER nor PJSIP_RESPONSE_HEADER can help with my usecase. I have already used pjsip session refresh to renegotiate media codecs during a call, but i've never tried adding an extraheader to the re-invite/update method. I will try this asap, thank you for the detailed answer. De : Users de la part de Gregory Massel Envoyé : jeudi 29 décembre 2022 20:28 À : users@lists.opensips.org Objet : Re: [OpenSIPS-Users] Media server communication with OpenSIPS You can send custom headers from Asterisk to OpenSIPS using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_HEADER If you need Asterisk to receive customer headers from OpenSIPS, you can read custom headers added by OpenSIPS to its 200 response using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_RESPONSE_HEADER If you are already midway through a call, you can force an UPDATE or re-INVITE using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_SEND_SESSION_REFRESH While I haven't personally tested, I'd expect that you should be able to signal custom information midway through a call with a Set(PJSIP_HEADER(add,X-MyHeader)=myvalue) followed by a Set(PJSIP_SEND_SESSION_REFRESH()=invite). This should then trigger a re-INVITE carrying the new X-MyHeader within it. Similarly, after triggering the a re-INVITE or UPDATE, you should then be able to do a Set(MYVAR=PJSIP_RESPONSE_HEADER(read,X-MyHeader)) to extract what OpenSIPS has sent back to Asterisk in its 200 response. Although more messy, there are also other tricks that you can use, including the CONNECTEDLINE function: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_CONNECTEDLINE e.g. If you did a Set(CONNECTEDLINE(name)=Information_I_wish_to_share_with_OpenSIPS), then this will, in one step, trigger an UPDATE or re-INVITE carrying the new connected party name in the SIP headers, detail which you can then extract from OpenSIPS. Similarly, you can also use this approach read information that OpenSIPS has sent to Asterisk, with a Set(MYVAR=CONNECTEDLINE(name)). I mention this in case, for whatever reason, the first approach (described above) doesn't work as planned, however, the first approach is definitely preferable as you can carry the information in your own X- header, making it far easier to send/receive multiple headers. On Tue, Dec 27, 2022 at 8:55 AM Wadii ELMAJDI | Evenmedia mailto:wa...@evenmedia.fr>> wrote: > > Hello, > > > I am using Asterisk as a media server behind OpenSIPS. > I need to send some additional data to opensips during the call or at the > latest during the hangup. For instance a custom hang up reason. > Asterisk does not allow neither to send a sequential request withing an > ongoing dialog, nor add headers to a BYE method. > The current solution I’m using is pushing a dialog var (dlg_push_var) using > Call-ID via (mi_http), but that means I rely on a curl http request during my > call… > Is there any better option for a direct communication between the two boxes > during a call ? without any intermediate server (Redis DB …) > > Thank you > > ___ > Users mailing list > Users@lists.opensips.org<mailto:Users@lists.opensips.org> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org<mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Media server communication with OpenSIPS
You can send custom headers from Asterisk to OpenSIPS using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_HEADER If you need Asterisk to receive customer headers from OpenSIPS, you can read custom headers added by OpenSIPS to its 200 response using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_RESPONSE_HEADER If you are already midway through a call, you can force an UPDATE or re-INVITE using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_SEND_SESSION_REFRESH While I haven't personally tested, I'd expect that you should be able to signal custom information midway through a call with a Set(PJSIP_HEADER(add,X-MyHeader)=myvalue) followed by a Set(PJSIP_SEND_SESSION_REFRESH()=invite). This should then trigger a re-INVITE carrying the new X-MyHeader within it. Similarly, after triggering the a re-INVITE or UPDATE, you should then be able to do a Set(MYVAR=PJSIP_RESPONSE_HEADER(read,X-MyHeader)) to extract what OpenSIPS has sent back to Asterisk in its 200 response. Although more messy, there are also other tricks that you can use, including the CONNECTEDLINE function: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_CONNECTEDLINE e.g. If you did a Set(CONNECTEDLINE(name)=Information_I_wish_to_share_with_OpenSIPS), then this will, in one step, trigger an UPDATE or re-INVITE carrying the new connected party name in the SIP headers, detail which you can then extract from OpenSIPS. Similarly, you can also use this approach read information that OpenSIPS has sent to Asterisk, with a Set(MYVAR=CONNECTEDLINE(name)). I mention this in case, for whatever reason, the first approach (described above) doesn't work as planned, however, the first approach is definitely preferable as you can carry the information in your own X- header, making it far easier to send/receive multiple headers. On Tue, Dec 27, 2022 at 8:55 AM Wadii ELMAJDI | Evenmedia wrote: > > Hello, > > > I am using Asterisk as a media server behind OpenSIPS. > I need to send some additional data to opensips during the call or at the latest during the hangup. For instance a custom hang up reason. > Asterisk does not allow neither to send a sequential request withing an ongoing dialog, nor add headers to a BYE method. > The current solution I’m using is pushing a dialog var (dlg_push_var) using Call-ID via (mi_http), but that means I rely on a curl http request during my call… > Is there any better option for a direct communication between the two boxes during a call ? without any intermediate server (Redis DB …) > > Thank you > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Media server communication with OpenSIPS
Hello, I must send more data than just a hangup code: a duration, a text value.. X-Asterisk-HangupCauseCode is only sent when a called is hung up after being answered. Plus, it isn’t really a custom header. It must be an int value. Same for Q.850 reason header. I will maintain my current solution then (the in-dialog request via mi_http) Thank you De : Users De la part de Brett Nemeroff Envoyé : mardi 27 décembre 2022 16:18 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] Media server communication with OpenSIPS If your UAC doesn't have the capabilities of sending it in the BYE or an in-dialog request, then you'll have to do something out of band like HTTPS, Redis, etc, which are not bad options. Have you looked into "X-Asterisk-HangupCauseCode" to see if that can help you out? On Tue, Dec 27, 2022 at 8:55 AM Wadii ELMAJDI | Evenmedia mailto:wa...@evenmedia.fr>> wrote: > > Hello, > > > I am using Asterisk as a media server behind OpenSIPS. > I need to send some additional data to opensips during the call or at the > latest during the hangup. For instance a custom hang up reason. > Asterisk does not allow neither to send a sequential request withing an > ongoing dialog, nor add headers to a BYE method. > The current solution I’m using is pushing a dialog var (dlg_push_var) using > Call-ID via (mi_http), but that means I rely on a curl http request during my > call… > Is there any better option for a direct communication between the two boxes > during a call ? without any intermediate server (Redis DB …) > > Thank you > > ___ > Users mailing list > Users@lists.opensips.org<mailto:Users@lists.opensips.org> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Media server communication with OpenSIPS
If your UAC doesn't have the capabilities of sending it in the BYE or an in-dialog request, then you'll have to do something out of band like HTTPS, Redis, etc, which are not bad options. Have you looked into "X-Asterisk-HangupCauseCode" to see if that can help you out? On Tue, Dec 27, 2022 at 8:55 AM Wadii ELMAJDI | Evenmedia < wa...@evenmedia.fr> wrote: > > Hello, > > > I am using Asterisk as a media server behind OpenSIPS. > I need to send some additional data to opensips during the call or at the latest during the hangup. For instance a custom hang up reason. > Asterisk does not allow neither to send a sequential request withing an ongoing dialog, nor add headers to a BYE method. > The current solution I’m using is pushing a dialog var (dlg_push_var) using Call-ID via (mi_http), but that means I rely on a curl http request during my call… > Is there any better option for a direct communication between the two boxes during a call ? without any intermediate server (Redis DB …) > > Thank you > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Media server communication with OpenSIPS
Hello, I am using Asterisk as a media server behind OpenSIPS. I need to send some additional data to opensips during the call or at the latest during the hangup. For instance a custom hang up reason. Asterisk does not allow neither to send a sequential request withing an ongoing dialog, nor add headers to a BYE method. The current solution I'm using is pushing a dialog var (dlg_push_var) using Call-ID via (mi_http), but that means I rely on a curl http request during my call... Is there any better option for a direct communication between the two boxes during a call ? without any intermediate server (Redis DB ...) Thank you ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
