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! Best regards, kokoska.rokoska