[Freeswitch-users] no RTP send during Voice Mail recording
kokoska.rokoska at post.cz
Mon Apr 20 11:21:45 PDT 2009
Thank you very much, Chris, for your reply!
Chris Chen napsal(a):
> 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,
I'm afraid (well, I'm pretty sure) non of them want to do it, because
they need very accurate billing and this is simpliest way how to do it -
kill calls without RTP i few seconds.
> 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.
Yes, I know. I have spent some years with * in the past (from "pre 1.0"
release if I remember correctly :-).
In my post I mean * ability to send faked audio during recording:
transmit_silence_during_record=yes option in asterisk.conf
> This works with one of my ITSP as they provide SIP trunking via *
None of my TSPs use Asterisk :-)
Around me there are much more popular Cirpacks and Phonets - due to
scalability, features, SS7 support etc...
> Hope this helps.
Thanks once more, Chris, for your interest!
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