[OpenSIPS-Users] load_balance not releasing resources

[Schneur Rosenberg]

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.
>>
>>
>>
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>>
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>>
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>>
>>
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