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
Anthony Minessale napsal(a):
> it's nothing to do with vad, it's simply how FS works.Thank you very much, Anthony, for explanation!
>
Yes, I understand. But can do nothing with it :-)
> 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.
>
Like I wrote - I should live with it.
> I understand it's not your fault, I am just letting you know.
>
Interesting variable name :-)
> 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.
>
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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