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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