[OpenSER-Users] Avoid Loops between Openser and Asterisk

[Adrian A]

If the INVITE comes in from Asterisk, OpenSER replies with 480 and lets
Asterisk deal with sending call to voicemail (e.g. dialstatus =
unavailable). This eliminates the loop because the INVITE does not come back
to Asterisk.

On Fri, Apr 11, 2008 at 12:33 AM, Iñaki Baz Castillo <ibc at in.ilimit.es>
wrote:

> El Thursday 10 April 2008 23:00:28 Adrian A escribió:
> > Here is how I solve this.
> >
> > In Asterisk, I add a SIP header just before sending it to OpenSER using
> the
> > function: SipAddHeader(P-hint: From Asterisk).
> >
> > Then, in OpenSER I have the following:
> >
> >                 # check to see if user is online
> >                  if (!lookup("location")) {
> >                         # SIP from Asterisk or not a call attempt
> >                         if (search("(P-hint):.From Asterisk") ||
> > !(method==INVITE)) { sl_send_reply("480","User Temporarily
> Unavailable");
> > exit;
> >                         } else {
> >                                 route(2);
> >                                  exit;
> >                         };
> >                 };
> >
> > route[2] {
> >         # Let Asterisk deal with voicemail/forward
> >         sl_send_reply("181", "Forward to Voicemail");
> >          prefix("vm");
> >         setflag(9);       # forward to Asterisk
> >         route(1);
> > }
>
> I can't understand why that should work, in fact Asterisk is receiving the
> same INVITE he sent (well, RURI modified but the Asterisk bug is that it
> doesn't recogniza a spiral).
>
>
> --
> Iñaki Baz Castillo
> ibc at in.ilimit.es
> _______________________________________________
> Users mailing list
> Users at lists.openser.org
> http://lists.openser.org/cgi-bin/mailman/listinfo/users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.kamailio.org/pipermail/users/attachments/20080411/3430e165/attachment.htm