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.rokoska@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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