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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org