We would have to code in a feature to purposely write silence back during a recording that does not currently exist. You could perhaps post it on the bounty section in jira.
On Tue, Feb 24, 2009 at 2:02 PM, <[email protected]> wrote: > no, I'm matching the incoming sip call via the destination number in my > public context and executing the javascript appliaction. This app directly > answers the call and records it until the user hangs up. > D- > > > ----- Original Message ----- > From: "Anthony Minessale" <[email protected]> > To: [email protected] > Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain > Subject: Re: [Freeswitch-users] Recording and outbound rtp > > is it during a bridged call? > > > On Tue, Feb 24, 2009 at 11:49 AM, Dan <[email protected]>wrote: > >> Hi, >> >> I have a small javascript application that accepts a call, retrieves some >> dtmf digits and then records the call to an icecast server. This works >> great. >> >> The problem I'm having is that when the call is being recorded freeswitch >> is no longer sending rtp packets back to the originating caller, in my case >> a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, >> since no voice data back is being generated. Unfortunately my Cisco gear >> has rtp inactivity timers set up to hang up a call after 3 minutes of no >> incoming rtp packets, this is a global setting that cannot be configured for >> a single dial peer. Does anyone have a suggestion to generate rtp packets >> every once in a while? I tried setting comfort noise which did not seem to >> send anything. I could try playing a empty/short wav file every minute or >> so but the javascript call session.record is blocking, would a traditional >> javascript timer and callback to play a wav file be my best bet or is there >> a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian >> etch. >> >> Thanks! >> Dan- >> >> _______________________________________________ >> 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 > > _______________________________________________ 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 > > _______________________________________________ > 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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