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
<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
_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
