[Freeswitch-users] Recording and outbound rtp

[freeswitch-users at digitaldan.com]

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" <anthony.minessale at gmail.com> 
To: freeswitch-users at lists.freeswitch.org 
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 < freeswitch-users at digitaldan.com > 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 
Freeswitch-users at lists.freeswitch.org 
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:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:213-799-1400 

_______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/e8956712/attachment-0002.html