[SR-Users] Dialog module showing calls that have already been terminated

[Phillman25 Kyriacou]

Hey Timo

Thanks for your email.
I apologise i never copied the config properly. I missed a  } to close the
if statement.
You can see that the route(WITHINDLG); is called for all requests from this
config.



       # MANAGE ALL DIALOGS
       #===================================================
        if (is_method("INVITE"))
 {
        if(is_method("INVITE") && !has_totag())
       {
           $dlg_ctx(timeout_route) = 12;
           $dlg_ctx(timeout_bye) = 1;
       }

          dlg_manage();

 }


         if(is_method("BYE|CANCEL"))

      {

         dlg_manage();


      }



        # per request initial checks
        route(REQINIT);

        # NAT detection
        route(NAT);

        # handle requests within SIP dialogs
        route(WITHINDLG);

        ### only initial requests (no To tag)

        # CANCEL processing
        if (is_method("CANCEL"))
        {
                if (t_check_trans())
                        t_relay();
                exit;
        }

        t_check_trans();

        # authentication
        route(AUTH);

        # record routing for dialog forming requests (in case they are
routed)
        # - remove preloaded route headers
        remove_hf("Route");
        if (is_method("INVITE|SUBSCRIBE"))
                record_route();



Thanks again for your help.

Phillip

On Tue, Sep 20, 2011 at 3:06 PM, Timo Reimann <timo.reimann at 1und1.de> wrote:

> Hey Phillip,
>
>
> On 20.09.2011 13:48, Phillman25 Kyriacou wrote:
> > Thanks for your email.
> >
> > Yes dlg_manage(); has to now be called on INVITE and BYE/CANCEL messages.
> > Where would i have to call loose_route()? Only on INVITE?
>
> On *all* in-dialog requests, i.e., all requests which contain a To tag.
> This may include Re-INVITEs too (but not initial INVITEs).
>
>
> > My configuration did not change between 3.1.2 and 3.1.5.
> >
> > Call flow example:
> > ==============
> >
> > Cisco PGW ===> Kamailio 3.1.5 ===> VOIP PROVIDER or ASTERISK PABX
> >
> >
> > The below is my configuration.
>
> Let's take a look at it:
>
>
> > #!KAMAILIO
> > #
>
> [...]
>
> >          if(is_method("BYE|CANCEL"))
> >
> >       {
> >
> >          dlg_manage();
> >
> >
> >
> >         # per request initial checks
> >         route(REQINIT);
> >
> >         # NAT detection
> >         route(NAT);
> >
> >         # handle requests within SIP dialogs
> >         route(WITHINDLG);
>
> [...]
>
> > # Handle requests within SIP dialogs
> > route[WITHINDLG] {
> >         if (has_totag()) {
> >                 # sequential request withing a dialog should
> >                 # take the path determined by record-routing
> >                 if (loose_route()) {
>
> [...]
>
>
> So route[WITHINDLG] contains the logic to track in-dialog requests.
> However, you seem to call that route only from BYE and CANCEL requests,
> missing out ACKs, which is likely the reason why things go wrong. So
> make sure you run all in-dialog requests through that route, and you
> should be fine (hopefully).
>
>
> Cheers,
>
> --Timo
>
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