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

[osiris123d]

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.

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