Hi Folks, Since I upgraded my asterisk box from 1.2.x to 1.4.x (1.4.10.1 now) I noticed some dead calls "apparently" running for more than 8 hours. I'm using rtptimeout=60 and rtpholdtimeout=120 and found some log messages like this:
chan_sip.c: 'SIP/XXX-085a9308' will not be disconnected in 61 seconds because it is directly bridged to another RTP stream I can kill that calls using 'soft hangup <channel>' but I'd like to know if its a new BUG introduced in 1.4.x releases and if possible, how to fix this? Thanks in advance. Rodrigo P. Telles _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
