4.1.3 supports SIP trunks - I would HIGHLY recommend you move to that. Anything over 4.0 supports SIP trunking.
-Greg On Thu, 2006-09-28 at 19:32 +0200, Yusuf wrote: > Hi, > > I recently had to hook up to Cisco Call Manager 4.1.3, and it only > supports H323. SO I used ooh323, and a strange thing happens. When a > Cisco IP user calls from his phone, the call gets sent from Call Manager > to Asterisk, but the phone will ring once only, then it seems asterisk > will drop the call, and int he debug it says: "stopped from reciving > frames from OOH323/cisco , bridging is being stopped". > > What is wrong? > > What RTP ports must I be using? > > thanks, > yusuf > > _______________________________________________ --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
