[OpenSIPS-Users] load_balance not releasing resources

[duane.larson at gmail.com]

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.



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