22 February 2007 12:22, Olle E Johansson wrote: > > 22 feb 2007 kl. 12.20 skrev Steve Langstaff: > > > Are the RTP timers applicable with canreinvite=yes ? > > how could we possibly check RTP if the RTP doesn't touch or > network card at all?
You can't. I realise. > The timers are only used when we have RTP streams going to > us. If the RTP stream is redirected, it's up to the end > points to hangup due to media failure. The endpoints have been rebooted, so they can't detect media failure (unless they have some persistent store of call state over a reboot!). > The way to solve this is to implement the SIP timer extension. I see there is a discussion of this on the bug tracker... http://bugs.digium.com/bug_view_page.php?bug_id=0000207 Looks like I'm going to be pushing the media through the server after all... _______________________________________________ --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
