[OpenSIPS-Users] Parallel Forking messes up Voicemail two-way audio

[Bogdan-Andrei Iancu]

Hi,

Let me first try to understand the scenario from SIP point of view.

In the first place, the call is sent to USER and PSTN (parallel). Now, 
the PSTN responds very slow and the USER (callee) decline the call, right ?

How the VM kicks in? serial forking?

Maybe a call flow or a trace will help a lot to understand your case.

Regards,
Bogdan

osiris123d wrote:
> I am wondering if anyone has run into this issue and how it might get fixed.
>
> I am testing a Hunt Group call where the user in the location table and a
> number out on the PSTN both get called at the same time since they both have
> the same Q value.  The parallel forking works just fine but due to the PSTN
> taking a little longer to respond to the invite the call to the location
> table user will always cancel before the call to the PSTN number.  Because
> of this I see that the call to the PSTN number is still going on when the
> Voicemail server picks up.  I think because the call to the PSTN user was
> still in process it messes up the Two-Way audio.  You can't hear the audio
> coming from the Voicemail server.  I know for a fact that my mediaproxy
> functions are set up correctly because on occasion it will work correctly. 
> Any idea how to fix this?  I tried the sleep() function in the failure
> route, but that didn't seem to help.
>
>   
-- 
Bogdan-Andrei Iancu
www.voice-system.ro