On Tuesday 03 April 2007 20:09, [EMAIL PROTECTED] wrote: > Charles Ulrich wrote: > > I have an Asterisk system deployed at a customer's site. It is > > connected to the outside world by a local SIP provider. When > > someone calls in through the trunk to leave a voicemail, Asterisk > > is not sending any RTP packets back through the trunk after the > > beep is played. This is fine and probably should be the expected > > behavior, except that after 30 seconds to a minute of not seeing > > any RTP traffic coming from the PBX, the trunk appears to make the > > faulty assumption that the PBX is gone and hangs up the call. > > Maybe this is what you need?: > > ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream > to keep NAT open ; (default is off - zero) > (in sip.conf, [general] section) > > Regards, > Philipp
That was exactly what I needed, thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
