Hi kokoska
Actually, you can request your VSP to set the rtptimeout or whatever
parameter in their SIP server to a reasonable value such as 300 seconds as 5
minutes should be enough for most standard business voice mail service,
otherwise you should wait for live calls instead of leaving voice messages.
In * they have the following setting which is default to 60 seconds if
nothing changed
rtptimeout=300 ; Terminate call if 60 seconds of no RTP or
RTCP activity
; on the audio channel
; when we're not on hold. This is to be able
to hangup
; a call in the case of a phone disappearing
from the net,
; like a powerloss or grandma tripping over
a cable.
This works with one of my ITSP as they provide SIP trunking via *
Hope this helps.
Chris
On Mon, Apr 20, 2009 at 12:48 PM, kokoska rokoska
<kokoska.roko...@post.cz>wrote:
>
> Anthony Minessale napsal(a):
> > it's nothing to do with vad, it's simply how FS works.
> >
>
> Thank you very much, Anthony, for explanation!
>
> > It's a waste to encode and send zeros into the channel while it's
> recording.
> > Also, It's unreasonable to have such a short timeout.
> >
>
> Yes, I understand. But can do nothing with it :-)
>
> > I understand it's not your fault, I am just letting you know.
> >
>
> Like I wrote - I should live with it.
>
> > It would be possible to add a patch to create a channel variable like
> > NDLB_waste_bandwidth_while_recording or something but it does not exist
> > today.
> >
>
> Interesting variable name :-)
> This will waste bandwidth, I'm sure, but will also save my life (from
> "not so happy" users). And from shame to go back to, I am ashamed to
> write it, "*" :-)
>
> Thanks once more, Anthony, for your help and useful informations!
>
>
> Best regards,
>
> kokoska.rokoska
>
>
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