Hi, It's really weired issue,I'm facing with asterisk-1.2.10 version. I see SIP call sessions stuck in asterisk for hours and then somehow get released. There happens to be an issue with BYE/CANCEL release msgs b/w sip entities. Has anyone faced this issue before also rtptimeout option given in sip.conf is not helping out.
Any suggestions? -AG x post to *-dev, *-users _______________________________________________ --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
