[OpenSIPS-Users] load_balance not releasing resources

[duane.larson at gmail.com]

Could you provide a sip trace of a call from INVITE to BYE? Also in your  
opensips config look and see where you have "404 Not here" configured.



On , Schneur Rosenberg <rosenberg11219 at gmail.com> wrote:
> In my case this is not relevant, because I'm calling the other phone

> through a DID and the did needs to go to asterisk to decide what to do

> with it, it can send it to a IVR which can later send it to Opensips

> etc. in any case I need to know why asterisk is not sending the BYE to

> the phone, and why opensips sends a not here when the BYE comes from a

> phone not on the system, in that case asterisk sends the BYE to

> opensips which sends a not here instead of sending it to the phone



> On Tue, Nov 15, 2011 at 2:06 AM, duane.larson at gmail.com> wrote:

> > If you want VM then you send to Asterks when the call times out (AKA the

> > callee doesn't pick up). We weren't talking about VM here. If you want  
> MOH

> > then that is a totally different beast. You would always have to send  
> the

> > calls to Asterisk and Asterisk would stay in the flow of the call. From  
> what

> > I read above it sounded like the following

> >

> > When I call from one phone on the system to another phone on the

> > same opensips, the phone sends a BYE to opensips which sends it to the

> > asterisk but the BYE never gets sent to the called phone.

> >

> > Sounds like Asterisk is not sending the BYE back to OpenSIPS because its

> > stated " opensips which sends it to the asterisk but the BYE never gets  
> sent

> > to the called phone."

> >

> >

> >

> >

> > On , Nick Khamis symack at gmail.com> wrote:

> >> On Mon, Nov 14, 2011 at 6:50 PM, duane.larson at gmail.com> wrote:

> >>

> >> > If two phones are registered with OpenSIPS and they call each other  
> why

> >>

> >> > would you send the SIP messages to Asterisk?

> >>

> >>

> >>

> >> Because "http://www.opensips.org/Resources/DocsTutAsterisk" said so! ;)

> >>

> >>

> >>

> >> > You need to set up route logic so that if two local users call each

> >> > other then

> >>

> >> > the asterisk boxes are kept out of the equation.

> >>

> >>

> >>

> >> Amazing idea! But what would happen to MOH, and VM?

> >>

> >>

> >>

> >> Nick.

> >>

> >>

> >>

> >> _______________________________________________

> >>

> >> Users mailing list

> >>

> >> Users at lists.opensips.org

> >>

> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

> >>

> >>

> > _______________________________________________

> > Users mailing list

> > Users at lists.opensips.org

> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users

> >

> >



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