it's nothing to do with vad, it's simply how FS works. 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.
I understand it's not your fault, I am just letting you know. 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. On Mon, Apr 20, 2009 at 11:03 AM, kokoska rokoska <[email protected]>wrote: > > > > kokoska.rokoska napsal(a): > > Hi all, > > > > I fall into trouble with voice mail. It looks like FreeSWITCH sends no > > RTP during Voice Mail recording and thus the calls from my TSPs are cut > > off in the middle of the recording due to lack of RTP activity (based on > > providers "tolerancy" it happens in 5 to 20 seconds). > > > > I tried to set VAD to "none" in all sofia profiles but it doesn't help. > > Are there any other settings I have to use to force FreeSWITCH to send > > RTP back (silence, CNG or what ever :-) during VM recording? > > > > BTW: I'm on current trunk. > > > > Hi all, > > until previous message I have tried all combinations of VAD settings and > VM recording format and still no luck: > using ngrep I can't see any RTP packetes going from FreeSWITCH during VM > recording => calls are dropped by my TSPs after few seconds. > > Could you, please, point me to some other direction where should I > experient? Or is it desired behaviour of FreeSWITCH? > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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